[svn-commits] mjordan: trunk r418019 - in /trunk: ./ addons/ apps/ channels/ channels/h323/...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Jul 4 08:26:55 CDT 2014
Author: mjordan
Date: Fri Jul 4 08:26:37 2014
New Revision: 418019
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=418019
Log:
Remove many deprecated modules
Billing records are fair,
To get paid is quite bright,
You should really use ODBC;
Good-bye cdr_sqlite.
Microsoft did once push H.323,
Hell, we all remember NetMeeting.
But try to compile chan_h323 now
And you will take quite a beating.
The XMPP and SIP war was fierce,
And in the distant fray
Was birthed res_jabber/chan_jingle;
But neither to stay.
For everyone did care and chase what Google professed.
"Free Internet Calling" was what devotees cried,
But Google did change the specs so often
That the developers were happy the day chan_gtalk died.
And then there was that odd application
Dedicated to the Polish tongue.
app_saycountpl was subsumed by Say;
One could say its bell was rung.
To read and parse a file from the dialplan
You could (I guess) use an application.
app_readfile did fill that purpose, but I think
A function is perhaps better in its creation.
Barging is rude, I'm not sure why we do it.
Inwardly, the caller will probably sigh.
But if you really must do it,
Don't use app_dahdibarge, use ChanSpy.
We all despise the sound of tinny robots
It makes our queues so cold.
To control such an abomination
It's better to not use Wait/SetMusicOnHold.
It's often nice to know properties of a channel
It makes our calls right
We have a nice function called CHANNEL
And so SIPCHANINFO is sent off into the night.
And now things get odd;
Apparently one could delimit with a colon
Properties from the SIPPEER function!
Commas are in; all others are done.
Finally, a word on pipes and commas.
We're sorry. We can't say it enough.
But those compatibility options in asterisk.conf;
To maintain them forever was just too tough.
This patch removes:
* cdr_sqlite
* chan_gtalk
* chan_jingle
* chan_h323
* res_jabber
* app_saycountpl
* app_readfile
* app_dahdibarge
It removes the following applications/functions:
* WaitMusicOnHold
* SetMusicOnHold
* SIPCHANINFO
It removes the colon delimiter from the SIPPEER function.
Finally, it also removes all compatibility options that were configurable from
asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems.
Review: https://reviewboard.asterisk.org/r/3698/
Removed:
trunk/addons/app_saycountpl.c
trunk/apps/app_dahdibarge.c
trunk/apps/app_readfile.c
trunk/channels/chan_gtalk.c
trunk/channels/chan_h323.c
trunk/channels/chan_jingle.c
trunk/channels/h323/
trunk/configs/gtalk.conf.sample
trunk/configs/jabber.conf.sample
trunk/configs/jingle.conf.sample
trunk/res/res_jabber.c
Modified:
trunk/CHANGES
trunk/UPGRADE.txt
trunk/addons/Makefile
trunk/channels/Makefile
trunk/channels/chan_sip.c
trunk/configs/asterisk.conf.sample
trunk/include/asterisk/options.h
trunk/main/asterisk.c
trunk/main/pbx.c
trunk/pbx/pbx_realtime.c
trunk/res/ael/pval.c
trunk/res/res_agi.c
trunk/res/res_musiconhold.c
trunk/utils/ael_main.c
trunk/utils/conf2ael.c
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Fri Jul 4 08:26:37 2014
@@ -12,6 +12,21 @@
--- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
------------------------------------------------------------------------------
+app_dahdibarge
+------------------
+ * This module was deprecated and has been removed. Users of app_dahdibarge
+ should use ChanSpy instead.
+
+app_readfile
+------------------
+ * This module was deprecated and has been removed. Users of app_readfile
+ should use func_env's FILE function instead.
+
+app_saycountpl
+------------------
+ * This module was deprecated and has been removed. Users of app_saycountpl
+ should use the Say family of applications.
+
AMI
------------------
* New DeviceStateChanged and PresenceStateChanged AMI events have been added.
@@ -30,6 +45,11 @@
* New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
enable manager control over PRI debugging levels and file output.
+cdr_sqlite
+-----------------
+ * This module was deprecated and has been removed. Users of cdr_sqlite
+ should use cdr_sqlite3_custom.
+
CEL
------------------
* The "bridge_technology" extra field key has been added to BRIDGE_ENTER
@@ -46,6 +66,30 @@
* Added several SS7 config option parameters described in
chan_dahdi.conf.sample.
+
+chan_gtalk
+------------------
+ * This module was deprecated and has been removed. Users of chan_gtalk
+ should use chan_motif.
+
+chan_h323
+------------------
+ * This module was deprecated and has been removed. Users of chan_h323
+ should use chan_ooh323.
+
+chan_jingle
+------------------
+ * This module was deprecated and has been removed. Users of chan_jingle
+ should use chan_motif.
+
+chan_sip
+------------------
+ * The SIPPEER dialplan function no longer supports using a colon as a
+ delimiter for parameters. The parameters for the function should be
+ delimited using a comma.
+
+ * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
+ of the function should use the CHANNEL function instead.
Core
------------------
@@ -79,6 +123,16 @@
------------------
* The JACK_HOOK function now supports audio with a sample rate higher than
8kHz.
+
+MusicOnHold
+------------------
+ * The SetMusicOnHold dialplan application was deprecated and has been removed.
+ Users of the application should use the CHANNEL function's musicclass
+ setting instead.
+
+ * The WaitMusicOnHold dialplan application was deprecated and has been
+ removed. Users of the application should use MusicOnHold with a duration
+ parameter instead.
Say
------------------
Modified: trunk/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/trunk/UPGRADE.txt?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/UPGRADE.txt (original)
+++ trunk/UPGRADE.txt Fri Jul 4 08:26:37 2014
@@ -43,6 +43,13 @@
directly. This change also includes a new script, refcounter.py, in the
contrib folder that will process the refs log file.
+ - The asterisk compatibility options in asterisk.conf have been removed.
+ These options enabled certain backwards compatibility features for
+ pbx_realtime, res_agi, and app_set that made their behaviour similar to
+ Asterisk 1.4. Users who used these backwards compatibility settings should
+ update their dialplans to use ',' instead of '|' as a delimiter, and should
+ use the Set dialplan application instead of the MSet dialplan application.
+
ARI:
- The ARI version has been changed from 1.0.0 to 1.1.0. This is to reflect
the backwards compatible changes listed below.
@@ -117,6 +124,9 @@
handler subroutine). In general, this is not the preferred default: this
causes extra CDRs to be generated for a channel in many common dialplans.
+ - The cdr_sqlite module was deprecated and has been removed. Users of this
+ module should use the cdr_sqlite3_custom module instead.
+
chan_dahdi:
- SS7 support now requires libss7 v2.0 or later.
@@ -124,6 +134,18 @@
deal with switches that don't send an inband progress indication in the
SETUP ACKNOWLEDGE message.
Default is now no.
+
+chan_gtalk
+ - This module was deprecated and has been removed. Users of chan_gtalk
+ should use chan_motif.
+
+chan_h323
+ - This module was deprecated and has been removed. Users of chan_h323
+ should use chan_ooh323.
+
+chan_jingle
+ - This module was deprecated and has been removed. Users of chan_jingle
+ should use chan_motif.
chan_pjsip:
- Added a 'force_avp' option to chan_pjsip which will force the usage of
@@ -138,6 +160,13 @@
chan_sip:
- Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
interoperability.
+
+ - The SIPPEER dialplan function no longer supports using a colon as a
+ delimiter for parameters. The parameters for the function should be
+ delimited using a comma.
+
+ - The SIPCHANINFO dialplan function was deprecated and has been removed. Users
+ of the function should use the CHANNEL function instead.
- Added a 'force_avp' option for chan_sip. When enabled this option will
cause the media transport in the offer or answer SDP to be 'RTP/AVP',
@@ -195,6 +224,15 @@
keep alive time between HTTP requests is configured in http.conf with the
session_keep_alive parameter.
+MusicOnHold
+ - The SetMusicOnHold dialplan application was deprecated and has been removed.
+ Users of the application should use the CHANNEL function's musicclass
+ setting instead.
+
+ - The WaitMusicOnHold dialplan application was deprecated and has been
+ removed. Users of the application should use MusicOnHold with a duration
+ parameter instead.
+
ODBC:
- The compatibility setting, allow_empty_string_in_nontext, has been removed.
Empty column values will be stored as empty strings during realtime updates.
@@ -241,6 +279,10 @@
- A new set of Alembic scripts has been added for CDR tables. This will create
a 'cdr' table with the default schema that Asterisk expects.
+res_jabber:
+ - This module was deprecated and has been removed. Users of this module should
+ use res_xmpp instead.
+
safe_asterisk:
- The safe_asterisk script was previously not installed on top of an existing
version. This caused bug-fixes in that script not to be deployed. If your
@@ -270,6 +312,5 @@
In such cases, it may be necessary to adjust this value.
Default is 100 ms.
-
===========================================================
===========================================================
Modified: trunk/addons/Makefile
URL: http://svnview.digium.com/svn/asterisk/trunk/addons/Makefile?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/addons/Makefile (original)
+++ trunk/addons/Makefile Fri Jul 4 08:26:37 2014
@@ -27,7 +27,6 @@
H323CFLAGS:=-Iooh323c/src -Iooh323c/src/h323
ALL_C_MODS:=app_mysql \
- app_saycountpl \
cdr_mysql \
chan_mobile \
chan_ooh323 \
Modified: trunk/channels/Makefile
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/Makefile?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/channels/Makefile (original)
+++ trunk/channels/Makefile Fri Jul 4 08:26:37 2014
@@ -15,40 +15,6 @@
MENUSELECT_CATEGORY=CHANNELS
MENUSELECT_DESCRIPTION=Channel Drivers
-ifeq ($(OSARCH),OpenBSD)
- PTLIB=-lpt
- H323LIB=-lh323
-endif
-
-ifeq ($(OSARCH),linux-gnu)
- PTLIB=-lpt_linux_x86_r
- H323LIB=-lh323_linux_x86_r
- CHANH323LIB=-ldl
-endif
-
-ifeq ($(OSARCH),FreeBSD)
- PTLIB=-lpt_FreeBSD_x86_r
- H323LIB=-lh323_FreeBSD_x86_r
- CHANH323LIB=-pthread
-endif
-
-ifeq ($(OSARCH),NetBSD)
- PTLIB=-lpt_NetBSD_x86_r
- H323LIB=-lh323_NetBSD_x86_r
-endif
-
-ifeq ($(wildcard h323/libchanh323.a),)
- MODULE_EXCLUDE += chan_h323
-endif
-
-ifndef OPENH323DIR
- OPENH323DIR=$(HOME)/openh323
-endif
-
-ifndef PWLIBDIR
- PWLIBDIR=$(HOME)/pwlib
-endif
-
all: _all
include $(ASTTOPDIR)/Makefile.moddir_rules
@@ -57,20 +23,12 @@
LIBS+= -lres_monitor.so -lres_features.so
endif
-ifneq ($(wildcard h323/Makefile.ast),)
-include h323/Makefile.ast
-endif
-
clean::
$(MAKE) -C misdn clean
rm -f dahdi/*.o dahdi/*.i
rm -f sip/*.o sip/*.i
rm -f iax2/*.o iax2/*.i
rm -f pjsip/*.o pjsip/*.i
- rm -f h323/libchanh323.a h323/Makefile.ast h323/*.o h323/*.dep
-
-dist-clean::
- rm -f h323/Makefile
$(if $(filter chan_iax2,$(EMBEDDED_MODS)),modules.link,chan_iax2.so): $(subst .c,.o,$(wildcard iax2/*.c))
$(subst .c,.o,$(wildcard iax2/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,chan_iax2)
@@ -91,20 +49,6 @@
$(if $(filter chan_dahdi,$(EMBEDDED_MODS)),modules.link,chan_dahdi.so): $(CHAN_DAHDI_OBJS)
$(CHAN_DAHDI_OBJS): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,chan_dahdi)
-ifneq ($(filter chan_h323,$(EMBEDDED_MODS)),)
-modules.link: h323/libchanh323.a
-else
-ifeq ($(OSARCH),linux-gnu)
-chan_h323.so: chan_h323.o h323/libchanh323.a
- $(ECHO_PREFIX) echo " [LD] $^ -> $@"
- $(CMD_PREFIX) $(CXX) $(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $(ASTLDFLAGS) $(SOLINK) -o $@ $< h323/libchanh323.a $(H323LDLIBS)
-else
-chan_h323.so: chan_h323.o h323/libchanh323.a
- $(ECHO_PREFIX) echo " [LD] $^ -> $@"
- $(CMD_PREFIX) $(CXX) $(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $(ASTLDFLAGS) $(SOLINK) -o $@ $< h323/libchanh323.a $(CHANH323LIB) -L$(PWLIBDIR)/lib $(PTLIB) -L$(OPENH323DIR)/lib $(H323LIB) -L/usr/lib -lcrypto -lssl -lexpat
-endif
-endif
-
chan_misdn.o: _ASTCFLAGS+=-Imisdn
misdn_config.o: _ASTCFLAGS+=-Imisdn
@@ -122,9 +66,3 @@
chan_usbradio.so: LIBS+=-lusb -lasound
chan_usbradio.so: _ASTCFLAGS+=-DNDEBUG
-h323/Makefile.ast:
- $(CMD_PREFIX) $(MAKE) -C h323 Makefile.ast
-
-h323/libchanh323.a: h323/Makefile.ast
- $(CMD_PREFIX) $(MAKE) -C h323 libchanh323.a
-
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Jul 4 08:26:37 2014
@@ -111,7 +111,7 @@
* \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
* \todo Save TCP/TLS sessions in registry
* If someone registers a SIPS uri, this forces us to set up a TLS connection back.
- * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
+ * \todo Add TCP/TLS information to function SIPPEER and CHANNEL function
* \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
* The tcpbindaddr config option should only be used to open ADDITIONAL ports
* So we should propably go back to
@@ -463,40 +463,6 @@
</syntax>
<description></description>
</function>
- <function name="SIPCHANINFO" language="en_US">
- <synopsis>
- Gets the specified SIP parameter from the current channel.
- </synopsis>
- <syntax>
- <parameter name="item" required="true">
- <enumlist>
- <enum name="peerip">
- <para>The IP address of the peer.</para>
- </enum>
- <enum name="recvip">
- <para>The source IP address of the peer.</para>
- </enum>
- <enum name="from">
- <para>The SIP URI from the <literal>From:</literal> header.</para>
- </enum>
- <enum name="uri">
- <para>The SIP URI from the <literal>Contact:</literal> header.</para>
- </enum>
- <enum name="useragent">
- <para>The Useragent header used by the peer.</para>
- </enum>
- <enum name="peername">
- <para>The name of the peer.</para>
- </enum>
- <enum name="t38passthrough">
- <para><literal>1</literal> if T38 is offered or enabled in this channel,
- otherwise <literal>0</literal>.</para>
- </enum>
- </enumlist>
- </parameter>
- </syntax>
- <description></description>
- </function>
<function name="CHECKSIPDOMAIN" language="en_US">
<synopsis>
Checks if domain is a local domain.
@@ -22390,15 +22356,11 @@
struct sip_peer *peer;
char *colname;
- if ((colname = strchr(data, ':'))) { /*! \todo Will be deprecated after 1.4 */
- static int deprecation_warning = 0;
+ if ((colname = strchr(data, ','))) {
*colname++ = '\0';
- if (deprecation_warning++ % 10 == 0)
- ast_log(LOG_WARNING, "SIPPEER(): usage of ':' to separate arguments is deprecated. Please use ',' instead.\n");
- } else if ((colname = strchr(data, ',')))
- *colname++ = '\0';
- else
+ } else {
colname = "ip";
+ }
if (!(peer = sip_find_peer(data, NULL, TRUE, FINDPEERS, FALSE, 0)))
return -1;
@@ -22493,77 +22455,6 @@
static struct ast_custom_function sippeer_function = {
.name = "SIPPEER",
.read = function_sippeer,
-};
-
-/*! \brief ${SIPCHANINFO()} Dialplan function - reads sip channel data */
-static int function_sipchaninfo_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
-{
- struct sip_pvt *p;
- static int deprecated = 0;
-
- *buf = 0;
-
- if (!chan) {
- ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
- return -1;
- }
-
- if (!data) {
- ast_log(LOG_WARNING, "This function requires a parameter name.\n");
- return -1;
- }
-
- ast_channel_lock(chan);
- if (!IS_SIP_TECH(ast_channel_tech(chan))) {
- ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
- ast_channel_unlock(chan);
- return -1;
- }
-
- if (deprecated++ % 20 == 0) {
- /* Deprecated in 1.6.1 */
- ast_log(LOG_WARNING, "SIPCHANINFO() is deprecated. Please transition to using CHANNEL().\n");
- }
-
- p = ast_channel_tech_pvt(chan);
-
- /* If there is no private structure, this channel is no longer alive */
- if (!p) {
- ast_channel_unlock(chan);
- return -1;
- }
-
- if (!strcasecmp(data, "peerip")) {
- ast_copy_string(buf, ast_sockaddr_stringify_addr(&p->sa), len);
- } else if (!strcasecmp(data, "recvip")) {
- ast_copy_string(buf, ast_sockaddr_stringify_addr(&p->recv), len);
- } else if (!strcasecmp(data, "from")) {
- ast_copy_string(buf, p->from, len);
- } else if (!strcasecmp(data, "uri")) {
- ast_copy_string(buf, p->uri, len);
- } else if (!strcasecmp(data, "useragent")) {
- ast_copy_string(buf, p->useragent, len);
- } else if (!strcasecmp(data, "peername")) {
- ast_copy_string(buf, p->peername, len);
- } else if (!strcasecmp(data, "t38passthrough")) {
- if ((p->t38.state == T38_DISABLED) || (p->t38.state == T38_REJECTED)) {
- ast_copy_string(buf, "0", len);
- } else { /* T38 is offered or enabled in this call */
- ast_copy_string(buf, "1", len);
- }
- } else {
- ast_channel_unlock(chan);
- return -1;
- }
- ast_channel_unlock(chan);
-
- return 0;
-}
-
-/*! \brief Structure to declare a dialplan function: SIPCHANINFO */
-static struct ast_custom_function sipchaninfo_function = {
- .name = "SIPCHANINFO",
- .read = function_sipchaninfo_read,
};
/*! \brief update redirecting information for a channel based on headers
@@ -34425,7 +34316,6 @@
/* Register dialplan functions */
ast_custom_function_register(&sip_header_function);
ast_custom_function_register(&sippeer_function);
- ast_custom_function_register(&sipchaninfo_function);
ast_custom_function_register(&checksipdomain_function);
/* Register manager commands */
@@ -34518,7 +34408,6 @@
ast_msg_tech_unregister(&sip_msg_tech);
/* Unregister dial plan functions */
- ast_custom_function_unregister(&sipchaninfo_function);
ast_custom_function_unregister(&sippeer_function);
ast_custom_function_unregister(&sip_header_function);
ast_custom_function_unregister(&checksipdomain_function);
Modified: trunk/configs/asterisk.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/asterisk.conf.sample?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/configs/asterisk.conf.sample (original)
+++ trunk/configs/asterisk.conf.sample Fri Jul 4 08:26:37 2014
@@ -95,8 +95,3 @@
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl
-
-[compat]
-pbx_realtime=1.6
-res_agi=1.6
-app_set=1.6
Modified: trunk/include/asterisk/options.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/options.h?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/include/asterisk/options.h (original)
+++ trunk/include/asterisk/options.h Fri Jul 4 08:26:37 2014
@@ -134,18 +134,6 @@
extern struct ast_flags ast_options;
-enum ast_compat_flags {
- AST_COMPAT_DELIM_PBX_REALTIME = (1 << 0),
- AST_COMPAT_DELIM_RES_AGI = (1 << 1),
- AST_COMPAT_APP_SET = (1 << 2),
-};
-
-#define ast_compat_pbx_realtime ast_test_flag(&ast_compat, AST_COMPAT_DELIM_PBX_REALTIME)
-#define ast_compat_res_agi ast_test_flag(&ast_compat, AST_COMPAT_DELIM_RES_AGI)
-#define ast_compat_app_set ast_test_flag(&ast_compat, AST_COMPAT_APP_SET)
-
-extern struct ast_flags ast_compat;
-
extern int option_verbose;
extern int ast_option_maxfiles; /*!< Max number of open file handles (files, sockets) */
extern int option_debug; /*!< Debugging */
Modified: trunk/main/asterisk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/asterisk.c?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/main/asterisk.c (original)
+++ trunk/main/asterisk.c Fri Jul 4 08:26:37 2014
@@ -317,7 +317,6 @@
/*! @{ */
struct ast_flags ast_options = { AST_DEFAULT_OPTIONS };
-struct ast_flags ast_compat = { 0 };
/*! Maximum active system verbosity level. */
int ast_verb_sys_level;
@@ -3646,20 +3645,7 @@
if (!ast_opt_remote) {
pbx_live_dangerously(live_dangerously);
}
- for (v = ast_variable_browse(cfg, "compat"); v; v = v->next) {
- float version;
- if (sscanf(v->value, "%30f", &version) != 1) {
- fprintf(stderr, "Compatibility version for option '%s' is not a number: '%s'\n", v->name, v->value);
- continue;
- }
- if (!strcasecmp(v->name, "app_set")) {
- ast_set2_flag(&ast_compat, version < 1.5 ? 1 : 0, AST_COMPAT_APP_SET);
- } else if (!strcasecmp(v->name, "res_agi")) {
- ast_set2_flag(&ast_compat, version < 1.5 ? 1 : 0, AST_COMPAT_DELIM_RES_AGI);
- } else if (!strcasecmp(v->name, "pbx_realtime")) {
- ast_set2_flag(&ast_compat, version < 1.5 ? 1 : 0, AST_COMPAT_DELIM_PBX_REALTIME);
- }
- }
+
ast_config_destroy(cfg);
}
Modified: trunk/main/pbx.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/pbx.c?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/main/pbx.c (original)
+++ trunk/main/pbx.c Fri Jul 4 08:26:37 2014
@@ -11465,10 +11465,6 @@
{
char *name, *value, *mydata;
- if (ast_compat_app_set) {
- return pbx_builtin_setvar_multiple(chan, data);
- }
-
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Set requires one variable name/value pair.\n");
return 0;
Modified: trunk/pbx/pbx_realtime.c
URL: http://svnview.digium.com/svn/asterisk/trunk/pbx/pbx_realtime.c?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/pbx/pbx_realtime.c (original)
+++ trunk/pbx/pbx_realtime.c Fri Jul 4 08:26:37 2014
@@ -303,7 +303,7 @@
struct ast_variable *var = realtime_common(context, exten, priority, data, MODE_MATCH);
if (var) {
- char *tmp="";
+ char *appdata_tmp = "";
char *app = NULL;
struct ast_variable *v;
@@ -311,31 +311,7 @@
if (!strcasecmp(v->name, "app"))
app = ast_strdupa(v->value);
else if (!strcasecmp(v->name, "appdata")) {
- if (ast_compat_pbx_realtime) {
- char *ptr;
- int in = 0;
- tmp = ast_alloca(strlen(v->value) * 2 + 1);
- for (ptr = tmp; *v->value; v->value++) {
- if (*v->value == ',') {
- *ptr++ = '\\';
- *ptr++ = ',';
- } else if (*v->value == '|' && !in) {
- *ptr++ = ',';
- } else {
- *ptr++ = *v->value;
- }
-
- /* Don't escape '|', meaning 'or', inside expressions ($[ ]) */
- if (v->value[0] == '[' && v->value[-1] == '$') {
- in++;
- } else if (v->value[0] == ']' && in) {
- in--;
- }
- }
- *ptr = '\0';
- } else {
- tmp = ast_strdupa(v->value);
- }
+ appdata_tmp = ast_strdupa(v->value);
}
}
ast_variables_destroy(var);
@@ -350,8 +326,8 @@
RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
appdata[0] = 0; /* just in case the substitute var func isn't called */
- if(!ast_strlen_zero(tmp))
- pbx_substitute_variables_helper(chan, tmp, appdata, sizeof(appdata) - 1);
+ if(!ast_strlen_zero(appdata_tmp))
+ pbx_substitute_variables_helper(chan, appdata_tmp, appdata, sizeof(appdata) - 1);
ast_verb(3, "Executing [%s@%s:%d] %s(\"%s\", \"%s\")\n",
ast_channel_exten(chan), ast_channel_context(chan), ast_channel_priority(chan),
term_color(tmp1, app, COLOR_BRCYAN, 0, sizeof(tmp1)),
Modified: trunk/res/ael/pval.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/ael/pval.c?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/res/ael/pval.c (original)
+++ trunk/res/ael/pval.c Fri Jul 4 08:26:37 2014
@@ -56,7 +56,6 @@
#endif
#include "asterisk/utils.h"
-extern struct ast_flags ast_compat;
extern int localized_pbx_load_module(void);
static char expr_output[2096];
@@ -3384,11 +3383,7 @@
for (first = 1; first >= 0; first--) {
switch_set = new_prio();
switch_set->type = AEL_APPCALL;
- if (!ast_compat_app_set) {
- switch_set->app = strdup("MSet");
- } else {
- switch_set->app = strdup("Set");
- }
+ switch_set->app = strdup("MSet");
/* Are we likely inside a gosub subroutine? */
if (!strcmp(mother_exten->name, "~~s~~") && first) {
/* If we're not actually within a gosub, this will fail, but the
@@ -3413,11 +3408,7 @@
for (first = 1; first >= 0; first--) {
switch_set = new_prio();
switch_set->type = AEL_APPCALL;
- if (!ast_compat_app_set) {
- switch_set->app = strdup("MSet");
- } else {
- switch_set->app = strdup("Set");
- }
+ switch_set->app = strdup("MSet");
/* Are we likely inside a gosub subroutine? */
if (!strcmp(exten->name, "~~s~~")) {
/* If we're not actually within a gosub, this will fail, but the
@@ -3453,11 +3444,7 @@
pr = new_prio();
pr->type = AEL_APPCALL;
snprintf(buf1, BUF_SIZE, "%s=$[%s]", p->u1.str, p->u2.val);
- if (!ast_compat_app_set) {
- pr->app = strdup("MSet");
- } else {
- pr->app = strdup("Set");
- }
+ pr->app = strdup("MSet");
remove_spaces_before_equals(buf1);
pr->appargs = strdup(buf1);
pr->origin = p;
@@ -3468,11 +3455,7 @@
pr = new_prio();
pr->type = AEL_APPCALL;
snprintf(buf1, BUF_SIZE, "LOCAL(%s)=$[%s]", p->u1.str, p->u2.val);
- if (!ast_compat_app_set) {
- pr->app = strdup("MSet");
- } else {
- pr->app = strdup("Set");
- }
+ pr->app = strdup("MSet");
remove_spaces_before_equals(buf1);
pr->appargs = strdup(buf1);
pr->origin = p;
@@ -3535,11 +3518,7 @@
for_test->goto_false = for_end;
for_loop->type = AEL_CONTROL1; /* simple goto */
for_end->type = AEL_APPCALL;
- if (!ast_compat_app_set) {
- for_init->app = strdup("MSet");
- } else {
- for_init->app = strdup("Set");
- }
+ for_init->app = strdup("MSet");
strcpy(buf2,p->u1.for_init);
remove_spaces_before_equals(buf2);
@@ -3600,11 +3579,7 @@
strncat(buf2,strp2+1, BUF_SIZE-strlen(strp2+1)-2);
strcat(buf2,"]");
for_inc->appargs = strdup(buf2);
- if (!ast_compat_app_set) {
- for_inc->app = strdup("MSet");
- } else {
- for_inc->app = strdup("Set");
- }
+ for_inc->app = strdup("MSet");
} else {
strp2 = p->u3.for_inc;
while (*strp2 && isspace(*strp2))
@@ -4489,11 +4464,7 @@
/* for each arg, set up a "Set" command */
struct ael_priority *np2 = new_prio();
np2->type = AEL_APPCALL;
- if (!ast_compat_app_set) {
- np2->app = strdup("MSet");
- } else {
- np2->app = strdup("Set");
- }
+ np2->app = strdup("MSet");
snprintf(buf,sizeof(buf),"LOCAL(%s)=${ARG%d}", lp->u1.str, argc++);
remove_spaces_before_equals(buf);
np2->appargs = strdup(buf);
Modified: trunk/res/res_agi.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_agi.c?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/res/res_agi.c (original)
+++ trunk/res/res_agi.c Fri Jul 4 08:26:37 2014
@@ -2767,24 +2767,7 @@
if (!(workaround = ast_test_flag(ast_channel_flags(chan), AST_FLAG_DISABLE_WORKAROUNDS))) {
ast_set_flag(ast_channel_flags(chan), AST_FLAG_DISABLE_WORKAROUNDS);
}
- if (ast_compat_res_agi && argc >= 3 && !ast_strlen_zero(argv[2])) {
- char *compat = ast_alloca(strlen(argv[2]) * 2 + 1), *cptr;
- const char *vptr;
- for (cptr = compat, vptr = argv[2]; *vptr; vptr++) {
- if (*vptr == ',') {
- *cptr++ = '\\';
- *cptr++ = ',';
- } else if (*vptr == '|') {
- *cptr++ = ',';
- } else {
- *cptr++ = *vptr;
- }
- }
- *cptr = '\0';
- res = pbx_exec(chan, app_to_exec, compat);
- } else {
- res = pbx_exec(chan, app_to_exec, argc == 2 ? "" : argv[2]);
- }
+ res = pbx_exec(chan, app_to_exec, argc == 2 ? "" : argv[2]);
if (!workaround) {
ast_clear_flag(ast_channel_flags(chan), AST_FLAG_DISABLE_WORKAROUNDS);
}
Modified: trunk/res/res_musiconhold.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_musiconhold.c?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/res/res_musiconhold.c (original)
+++ trunk/res/res_musiconhold.c Fri Jul 4 08:26:37 2014
@@ -96,36 +96,6 @@
Returns <literal>0</literal> when done, <literal>-1</literal> on hangup.</para>
<para>This application does not automatically answer and should be preceeded by
an application such as Answer() or Progress().</para>
- </description>
- </application>
- <application name="WaitMusicOnHold" language="en_US">
- <synopsis>
- Wait, playing Music On Hold.
- </synopsis>
- <syntax>
- <parameter name="delay" required="true" />
- </syntax>
- <description>
- <para> !!! DEPRECATED. Use MusicOnHold instead !!!</para>
- <para>Plays hold music specified number of seconds. Returns <literal>0</literal> when done,
- or <literal>-1</literal> on hangup. If no hold music is available, the delay will still occur
- with no sound.</para>
- <para> !!! DEPRECATED. Use MusicOnHold instead !!!</para>
- </description>
- </application>
- <application name="SetMusicOnHold" language="en_US">
- <synopsis>
- Set default Music On Hold class.
- </synopsis>
- <syntax>
- <parameter name="class" required="yes" />
- </syntax>
- <description>
- <para>!!! DEPRECATED. USe Set(CHANNEL(musicclass)=...) instead !!!</para>
- <para>Sets the default class for music on hold for a given channel.
- When music on hold is activated, this class will be used to select which
- music is played.</para>
- <para>!!! DEPRECATED. USe Set(CHANNEL(musicclass)=...) instead !!!</para>
</description>
</application>
<application name="StartMusicOnHold" language="en_US">
@@ -153,8 +123,6 @@
***/
static const char play_moh[] = "MusicOnHold";
-static const char wait_moh[] = "WaitMusicOnHold";
-static const char set_moh[] = "SetMusicOnHold";
static const char start_moh[] = "StartMusicOnHold";
static const char stop_moh[] = "StopMusicOnHold";
@@ -862,46 +830,6 @@
return res;
}
-static int wait_moh_exec(struct ast_channel *chan, const char *data)
-{
- static int deprecation_warning = 0;
- int res;
-
- if (!deprecation_warning) {
- deprecation_warning = 1;
- ast_log(LOG_WARNING, "WaitMusicOnHold application is deprecated and will be removed. Use MusicOnHold with duration parameter instead\n");
- }
-
- if (!data || !atoi(data)) {
- ast_log(LOG_WARNING, "WaitMusicOnHold requires an argument (number of seconds to wait)\n");
- return -1;
- }
- if (ast_moh_start(chan, NULL, NULL)) {
- ast_log(LOG_WARNING, "Unable to start music on hold for %d seconds on channel %s\n", atoi(data), ast_channel_name(chan));
- return 0;
- }
- res = ast_safe_sleep(chan, atoi(data) * 1000);
- ast_moh_stop(chan);
- return res;
-}
-
-static int set_moh_exec(struct ast_channel *chan, const char *data)
-{
- static int deprecation_warning = 0;
-
- if (!deprecation_warning) {
- deprecation_warning = 1;
- ast_log(LOG_WARNING, "SetMusicOnHold application is deprecated and will be removed. Use Set(CHANNEL(musicclass)=...) instead\n");
- }
-
- if (ast_strlen_zero(data)) {
- ast_log(LOG_WARNING, "SetMusicOnHold requires an argument (class)\n");
- return -1;
- }
- ast_channel_musicclass_set(chan, data);
- return 0;
-}
-
static int start_moh_exec(struct ast_channel *chan, const char *data)
{
char *parse;
@@ -2009,10 +1937,6 @@
ast_register_atexit(ast_moh_destroy);
ast_cli_register_multiple(cli_moh, ARRAY_LEN(cli_moh));
if (!res)
- res = ast_register_application_xml(wait_moh, wait_moh_exec);
- if (!res)
- res = ast_register_application_xml(set_moh, set_moh_exec);
- if (!res)
res = ast_register_application_xml(start_moh, start_moh_exec);
if (!res)
res = ast_register_application_xml(stop_moh, stop_moh_exec);
@@ -2058,8 +1982,6 @@
ast_moh_destroy();
res = ast_unregister_application(play_moh);
- res |= ast_unregister_application(wait_moh);
- res |= ast_unregister_application(set_moh);
res |= ast_unregister_application(start_moh);
res |= ast_unregister_application(stop_moh);
ast_cli_unregister_multiple(cli_moh, ARRAY_LEN(cli_moh));
Modified: trunk/utils/ael_main.c
URL: http://svnview.digium.com/svn/asterisk/trunk/utils/ael_main.c?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/utils/ael_main.c (original)
+++ trunk/utils/ael_main.c Fri Jul 4 08:26:37 2014
@@ -36,8 +36,6 @@
void ast_register_file_version(const char *file, const char *version) { }
void ast_unregister_file_version(const char *file) { }
#endif
-
-struct ast_flags ast_compat = { 7 };
/*** MODULEINFO
<depend>res_ael_share</depend>
Modified: trunk/utils/conf2ael.c
URL: http://svnview.digium.com/svn/asterisk/trunk/utils/conf2ael.c?view=diff&rev=418019&r1=418018&r2=418019
==============================================================================
--- trunk/utils/conf2ael.c (original)
+++ trunk/utils/conf2ael.c Fri Jul 4 08:26:37 2014
@@ -56,7 +56,6 @@
#include "asterisk/pval.h"
#include "asterisk/extconf.h"
-struct ast_flags ast_compat = { 7 };
const char *ast_config_AST_CONFIG_DIR = "/etc/asterisk"; /* placeholder */
void get_start_stop(unsigned int *word, int bitsperword, int totalbits, int *start, int *end);
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