[asterisk-commits] kpfleming: trunk r157706 - in /trunk: ./ apps/ include/asterisk/ res/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Nov 19 06:42:36 CST 2008
Author: kpfleming
Date: Wed Nov 19 06:42:19 2008
New Revision: 157706
URL: http://svn.digium.com/view/asterisk?view=rev&rev=157706
Log:
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
Added:
trunk/UPGRADE-1.6.txt
- copied unchanged from r157705, team/kpfleming/agi_multiple-api-fix/UPGRADE-1.6.txt
Modified:
trunk/UPGRADE.txt
trunk/apps/app_stack.c
trunk/include/asterisk/agi.h
trunk/res/res_agi.c
Modified: trunk/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/trunk/UPGRADE.txt?view=diff&rev=157706&r1=157705&r2=157706
==============================================================================
--- trunk/UPGRADE.txt (original)
+++ trunk/UPGRADE.txt Wed Nov 19 06:42:19 2008
@@ -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
-=========================================================
+=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
+===========================================================
-AEL:
+From 1.6.0.1 to 1.6.0.2 or later, or 1.6.1 or later:
-* 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 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 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 at 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.
+* 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.
Modified: trunk/apps/app_stack.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_stack.c?view=diff&rev=157706&r1=157705&r2=157706
==============================================================================
--- trunk/apps/app_stack.c (original)
+++ trunk/apps/app_stack.c Wed Nov 19 06:42:19 2008
@@ -491,11 +491,11 @@
/* 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 @@
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 @@
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;
}
Modified: trunk/include/asterisk/agi.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/agi.h?view=diff&rev=157706&r1=157705&r2=157706
==============================================================================
--- trunk/include/asterisk/agi.h (original)
+++ trunk/include/asterisk/agi.h Wed Nov 19 06:42:19 2008
@@ -66,11 +66,53 @@
#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)
}
Modified: trunk/res/res_agi.c
URL: http://svn.digium.com/view/asterisk/trunk/res/res_agi.c?view=diff&rev=157706&r1=157705&r2=157706
==============================================================================
--- trunk/res/res_agi.c (original)
+++ trunk/res/res_agi.c Wed Nov 19 06:42:19 2008
@@ -304,7 +304,7 @@
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 @@
}
}
- 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 @@
/* 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 @@
/* 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
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 @@
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 @@
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 @@
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 @@
/* 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 @@
}
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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) {
[... 397 lines stripped ...]
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