Matt! I thank you oh our leader great For this message following the farewell of a lot of old crap, old mate, which no one longer could sell.
But please don't change the commit rules! /O ;-) On 04 Jul 2014, at 15:26, SVN commits to the Digium repositories <[email protected]> wrote: > 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); > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > svn-commits mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/svn-commits -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
