diff options
author | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-11-19 12:42:19 +0000 |
---|---|---|
committer | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-11-19 12:42:19 +0000 |
commit | 7ad42d39db3d8c1e821bf3e9e48053d305b4a117 (patch) | |
tree | b4c66fb5a34118b5d64f89dd91dfe1ce7393ea42 | |
parent | 421ba2499605754b0b9b46b4c61511dd0cb5adea (diff) |
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/trunk@157706 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | UPGRADE-1.6.txt | 311 | ||||
-rw-r--r-- | UPGRADE.txt | 331 | ||||
-rw-r--r-- | apps/app_stack.c | 14 | ||||
-rw-r--r-- | include/asterisk/agi.h | 48 | ||||
-rw-r--r-- | res/res_agi.c | 281 |
5 files changed, 544 insertions, 441 deletions
diff --git a/UPGRADE-1.6.txt b/UPGRADE-1.6.txt new file mode 100644 index 000000000..3b7db9e4d --- /dev/null +++ b/UPGRADE-1.6.txt @@ -0,0 +1,311 @@ +========================================================= +=== 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 +========================================================= + +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. + +* The following core commands dealing with dialplan has been deprecated: 'core + show globals', 'core set global' and 'core set chanvar'. Use the equivalent + 'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar' + instead. + +* The silencethreshold used for various applications is now settable via a + centralized config option in dsp.conf. + +* The logical value of spaces immediately preceding a standalone 0 previously + evaluated to true. It now evaluates to false. This has confused a good + many people in the past (typically because they failed to realize the space + had any significance). Since this violates the Principle of Least Surprise, + it has been changed. + +* The default console now will use colors according to the default background + color, instead of forcing the background color to black. If you are using a + light colored background for your console, you may wish to use the option + flag '-W' to present better color choices for the various messages. However, + if you'd prefer the old method of forcing colors to white text on a black + background, the compatiblity option -B is provided for this purpose. + +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. + +* The externnotify script should accept an additional (last) parameter + containing the number of urgent messages in the INBOX. + +Applications: + +* SendImage() no longer hangs up the channel on transmission error or on + another type of error; in those cases, a FAILURE status is stored in + SENDIMAGESTATUS and dialplan execution continues. The possible return values + stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and UNSUPPORTED. ('OK' has + been replaced with 'SUCCESS', and 'NOSUPPORT' has been replaced with + 'UNSUPPORTED'). This change makes the SendImage application more consistent + with other 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. + +* While app_directory has always relied on having a voicemail.conf or users.conf file + correctly set up, it now is dependent on app_voicemail being compiled as well. + +* 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. + +* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(), + and you should start using that function instead for retrieving information about + the channel in a technology-agnostic way. + +* 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. + +* If you have any third party modules which use a config file variable whose + name ends in a '+', please note that the append capability added to this + version may now conflict with that variable naming scheme. An easy + workaround is to ensure that a space occurs between the '+' and the '=', + to differentiate your variable from the append operator. This potential + conflict is unlikely, but is documented here to be thorough. + +* skinny.conf now has seperate sections for lines and devices. + Please have a look at configs/skinny.conf.sample and update + your skinny.conf. + +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. + +Queues: + +* New queue log events ADDMEMBER and REMOVEMEMBER have been added. Also, a + new value has been added to the TRANSFER event that indicates the caller's + original position in the queue they are being transfered from. + +* Prior to Asterisk 1.6.2, queue names were treated in a case-sensitive + manner, meaning that queues with names like "sales" and "sALeS" would + be seen as unique queues. The parsing logic has changed to use case- + insensitive comparisons now when originally hashing based on queue + names, meaning that now the two queues mentioned as examples earlier + will be seen as having the same name. + +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 209ebd50d..ef13c46ac 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -1,312 +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. - -* The following core commands dealing with dialplan has been deprecated: 'core - show globals', 'core set global' and 'core set chanvar'. Use the equivalent - 'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar' - instead. - -* The silencethreshold used for various applications is now settable via a - centralized config option in dsp.conf. - -* The logical value of spaces immediately preceding a standalone 0 previously - evaluated to true. It now evaluates to false. This has confused a good - many people in the past (typically because they failed to realize the space - had any significance). Since this violates the Principle of Least Surprise, - it has been changed. - -* The default console now will use colors according to the default background - color, instead of forcing the background color to black. If you are using a - light colored background for your console, you may wish to use the option - flag '-W' to present better color choices for the various messages. However, - if you'd prefer the old method of forcing colors to white text on a black - background, the compatiblity option -B is provided for this purpose. - -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. - -* The externnotify script should accept an additional (last) parameter - containing the number of urgent messages in the INBOX. - -Applications: - -* SendImage() no longer hangs up the channel on transmission error or on - another type of error; in those cases, a FAILURE status is stored in - SENDIMAGESTATUS and dialplan execution continues. The possible return values - stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and UNSUPPORTED. ('OK' has - been replaced with 'SUCCESS', and 'NOSUPPORT' has been replaced with - 'UNSUPPORTED'). This change makes the SendImage application more consistent - with other 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. - -* While app_directory has always relied on having a voicemail.conf or users.conf file - correctly set up, it now is dependent on app_voicemail being compiled as well. - -* 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. - -* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(), - and you should start using that function instead for retrieving information about - the channel in a technology-agnostic way. - -* 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. - -* If you have any third party modules which use a config file variable whose - name ends in a '+', please note that the append capability added to this - version may now conflict with that variable naming scheme. An easy - workaround is to ensure that a space occurs between the '+' and the '=', - to differentiate your variable from the append operator. This potential - conflict is unlikely, but is documented here to be thorough. - -* skinny.conf now has seperate sections for lines and devices. - Please have a look at configs/skinny.conf.sample and update - your skinny.conf. - -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. - -Queues: - -* New queue log events ADDMEMBER and REMOVEMEMBER have been added. Also, a - new value has been added to the TRANSFER event that indicates the caller's - original position in the queue they are being transfered from. - -* Prior to Asterisk 1.6.2, queue names were treated in a case-sensitive - manner, meaning that queues with names like "sales" and "sALeS" would - be seen as unique queues. The parsing logic has changed to use case- - insensitive comparisons now when originally hashing based on queue - names, meaning that now the two queues mentioned as examples earlier - will be seen as having the same name. - -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 or later, or 1.6.1 or later: + +* 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 45db41d05..cebe161af 100644 --- a/apps/app_stack.c +++ b/apps/app_stack.c @@ -491,11 +491,11 @@ 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; } @@ -506,7 +506,7 @@ 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; } @@ -540,19 +540,19 @@ 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/include/asterisk/agi.h b/include/asterisk/agi.h index 48de8c954..c92ab3c66 100644 --- a/include/asterisk/agi.h +++ b/include/asterisk/agi.h @@ -66,11 +66,53 @@ 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 6344cfaff..c3683b0d5 100644 --- a/res/res_agi.c +++ b/res/res_agi.c @@ -304,7 +304,7 @@ 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; @@ -771,7 +771,7 @@ 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); @@ -782,7 +782,7 @@ 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; @@ -911,40 +911,40 @@ 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[]) @@ -955,7 +955,7 @@ 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; } @@ -968,7 +968,7 @@ 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; } @@ -987,7 +987,7 @@ 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; } @@ -1000,14 +1000,14 @@ 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; } - ast_agi_fdprintf(chan, agi->fd, "200 result=%d (hangup)\n", res); + ast_agi_send(agi->fd, chan, "200 result=%d (hangup)\n", res); return RESULT_FAILURE; } @@ -1020,10 +1020,10 @@ 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; } @@ -1048,9 +1048,9 @@ static int handle_tddmode(struct ast_channel *chan, AGI *agi, int argc, char *ar } 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; } @@ -1067,7 +1067,7 @@ static int handle_sendimage(struct ast_channel *chan, AGI *agi, int argc, char * 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; } @@ -1102,7 +1102,7 @@ static int handle_controlstreamfile(struct ast_channel *chan, AGI *agi, int argc res = ast_control_streamfile(chan, argv[3], fwd, rev, stop, suspend, 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; } @@ -1124,7 +1124,7 @@ 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; } @@ -1152,7 +1152,7 @@ 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; } @@ -1179,7 +1179,7 @@ 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; } @@ -1217,7 +1217,7 @@ 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; } @@ -1237,7 +1237,7 @@ 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; } @@ -1253,7 +1253,7 @@ 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; } @@ -1267,7 +1267,7 @@ 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; } @@ -1282,7 +1282,7 @@ 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; } @@ -1297,7 +1297,7 @@ 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; } @@ -1331,7 +1331,7 @@ 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; } @@ -1345,7 +1345,7 @@ 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; } @@ -1368,11 +1368,11 @@ 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; } @@ -1382,7 +1382,7 @@ 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; } @@ -1391,7 +1391,7 @@ 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; } @@ -1408,7 +1408,7 @@ 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; } @@ -1483,12 +1483,12 @@ 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; @@ -1508,14 +1508,14 @@ 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); @@ -1530,7 +1530,7 @@ 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) @@ -1575,7 +1575,7 @@ 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); } @@ -1605,7 +1605,7 @@ static int handle_autohangup(struct ast_channel *chan, AGI *agi, int argc, char whentohangup.tv_usec = (timeout - whentohangup.tv_sec) * 1000000.0; } ast_channel_setwhentohangup_tv(chan, whentohangup); - ast_agi_fdprintf(chan, agi->fd, "200 result=0\n"); + ast_agi_send(agi->fd, chan, "200 result=0\n"); return RESULT_SUCCESS; } @@ -1616,7 +1616,7 @@ 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 */ @@ -1624,12 +1624,12 @@ 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; @@ -1671,7 +1671,7 @@ 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; @@ -1694,7 +1694,7 @@ 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; } @@ -1703,18 +1703,18 @@ 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; @@ -1726,7 +1726,7 @@ 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; } @@ -1746,9 +1746,9 @@ 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; } @@ -1767,9 +1767,9 @@ 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); @@ -1788,7 +1788,7 @@ 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; } @@ -1802,9 +1802,9 @@ 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; } @@ -1816,7 +1816,7 @@ 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; } @@ -1827,7 +1827,7 @@ 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; } @@ -1842,7 +1842,7 @@ 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; } @@ -1877,7 +1877,7 @@ 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; } @@ -1887,7 +1887,7 @@ 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; } @@ -1895,14 +1895,14 @@ 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; } @@ -1915,12 +1915,12 @@ 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; } @@ -1930,9 +1930,9 @@ 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; @@ -1944,14 +1944,14 @@ 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; } @@ -1962,14 +1962,14 @@ 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; } @@ -1980,14 +1980,14 @@ 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; } @@ -1998,14 +1998,14 @@ 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; } @@ -2045,7 +2045,7 @@ 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; } @@ -2059,7 +2059,7 @@ 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; } @@ -2164,13 +2164,13 @@ 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; @@ -2178,7 +2178,7 @@ 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; } @@ -2545,21 +2545,50 @@ 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++) - ast_agi_register(mod, cmd + i); + for (i = 0; i < len; i++) { + if (ast_agi_register(mod, cmd + i) == 1) { + x++; + continue; + } + + /* 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) @@ -2701,9 +2730,9 @@ 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 */ @@ -2715,7 +2744,7 @@ 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" @@ -2724,7 +2753,7 @@ 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" @@ -3195,7 +3224,10 @@ 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"); @@ -3205,7 +3237,10 @@ 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); |