[asterisk-dev] [svn-commits] mjordan: trunk r418019 - in /trunk: ./ addons/ apps/ channels/ channels/h323/...

Olle E. Johansson oej at edvina.net
Fri Jul 4 08:49:51 CDT 2014


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 <svn-commits at lists.digium.com> 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 --
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