issue #2720

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7034 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ChangeLog

 2005-11-08  Kevin P. Fleming  <kpfleming@digium.com>
-	
+
+	* pbx.c (pbx_builtin_pushvar_helper): add new API function for setting variables that can exist multiple times (issue #2720)
+	* apps/Makefile (APPS): add app_stack (issue #2720)
+	* apps/app_stack.c: new applications (issue #2720)
+
 	* apps/app_meetme.c: fix two audio delay problems related to using non-Zap channels in conferences (issues #3599 and #4252)
 	* configs/meetme.conf.sample: add documentation of new 'audiobuffers' setting to control buffering on incoming audio from non-Zap channels
 

apps/Makefile

@@ -29,7 +29,7 @@ APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
      app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
      app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so \
      app_dictate.so app_externalivr.so app_directed_pickup.so \
-     app_mixmonitor.so
+     app_mixmonitor.so app_stack.so
 
 #
 # Obsolete things...

apps/app_stack.c

+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (c) 2004-2005 Tilghman Lesher <app_stack_v002@the-tilghman.com>.
+ *
+ * This code is released by the author with no restrictions on usage.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Stack applications Gosub, Return, etc.
+ * 
+ * \ingroup applications
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+
+#include "asterisk/options.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/chanvars.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/config.h"
+
+#define STACKVAR	"~GOSUB~STACK~"
+
+static const char *tdesc = "Stack Routines";
+
+static const char *app_gosub = "Gosub";
+static const char *app_gosubif = "GosubIf";
+static const char *app_return = "Return";
+static const char *app_pop = "StackPop";
+
+static const char *gosub_synopsis = "Jump to label, saving return address";
+static const char *gosubif_synopsis = "Jump to label, saving return address";
+static const char *return_synopsis = "Return from gosub routine";
+static const char *pop_synopsis = "Remove one address from gosub stack";
+
+static const char *gosub_descrip =
+"Gosub([[context|]exten|]priority)\n"
+"  Jumps to the label specified, saving the return address.\n"
+"  Returns 0 if the label exists or -1 otherwise.\n";
+static const char *gosubif_descrip =
+"Gosub(condition?labeliftrue[:labeliffalse])\n"
+"  If the condition is true, then jump to labeliftrue.  If false, jumps to\n"
+"labeliffalse, if specified.  In either case, a jump saves the return point\n"
+"in the dialplan, to be returned to with a Return.\n"
+"  Returns 0 if the label exists or -1 otherwise.\n";
+static const char *return_descrip =
+"Return()\n"
+"  Jumps to the last label in the stack, removing it.\n"
+"  Returns 0 if there's a label in the stack or -1 otherwise.\n";
+static const char *pop_descrip =
+"StackPop()\n"
+"  Removes last label in the stack, discarding it.\n"
+"  Always returns 0, even if the stack is empty.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int pop_exec(struct ast_channel *chan, void *data)
+{
+	pbx_builtin_setvar_helper(chan, STACKVAR, NULL);
+
+	return 0;
+}
+
+static int return_exec(struct ast_channel *chan, void *data)
+{
+	char *label = pbx_builtin_getvar_helper(chan, STACKVAR);
+
+	if (ast_strlen_zero(label)) {
+		ast_log(LOG_ERROR, "Return without Gosub: stack is empty\n");
+		return -1;
+	} else if (ast_parseable_goto(chan, label)) {
+		ast_log(LOG_WARNING, "No next statement after Gosub?\n");
+		return -1;
+	}
+
+	pbx_builtin_setvar_helper(chan, STACKVAR, NULL);
+	return 0;
+}
+
+static int gosub_exec(struct ast_channel *chan, void *data)
+{
+	char newlabel[AST_MAX_EXTENSION * 2 + 3 + 11];
+	struct localuser *u;
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_ERROR, "%s requires an argument: %s([[context|]exten|]priority)\n", app_gosub, app_gosub);
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+	snprintf(newlabel, sizeof(newlabel), "%s|%s|%d", chan->context, chan->exten, chan->priority + 1);
+
+	if (ast_parseable_goto(chan, data)) {
+		LOCAL_USER_REMOVE(u);
+		return -1;
+	}
+
+	pbx_builtin_pushvar_helper(chan, STACKVAR, newlabel);
+	LOCAL_USER_REMOVE(u);
+
+	return 0;
+}
+
+static int gosubif_exec(struct ast_channel *chan, void *data)
+{
+	struct localuser *u;
+	char *condition="", *label1, *label2, *args;
+	int res=0;
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "GosubIf requires an argument\n");
+		return 0;
+	}
+
+	args = ast_strdupa((char *)data);
+	if (!args) {
+		ast_log(LOG_ERROR, "Out of memory\n");
+		return -1;
+	}
+
+	LOCAL_USER_ADD(u);
+
+	condition = strsep(&args, "?");
+	label1 = strsep(&args, ":");
+	label2 = args;
+
+	if (ast_true(condition)) {
+		if (label1) {
+			res = gosub_exec(chan, label1);
+		}
+	} else if (label2) {
+		res = gosub_exec(chan, label2);
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	ast_unregister_application(app_return);
+	ast_unregister_application(app_pop);
+	ast_unregister_application(app_gosubif);
+	ast_unregister_application(app_gosub);
+
+	STANDARD_HANGUP_LOCALUSERS;
+
+	return 0;
+}
+
+int load_module(void)
+{
+	ast_register_application(app_pop, pop_exec, pop_synopsis, pop_descrip);
+	ast_register_application(app_return, return_exec, return_synopsis, return_descrip);
+	ast_register_application(app_gosubif, gosubif_exec, gosubif_synopsis, gosubif_descrip);
+	ast_register_application(app_gosub, gosub_exec, gosub_synopsis, gosub_descrip);
+
+	return 0;
+}
+
+char *description(void)
+{
+	return (char *) tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+
+	STANDARD_USECOUNT(res);
+
+	return res;
+}
+
+char *key()
+{
+	return ASTERISK_GPL_KEY;
+}

include/asterisk/pbx.h

@@ -606,6 +606,7 @@ struct ast_sw *ast_walk_context_switches(struct ast_context *con, struct ast_sw
 
 int pbx_builtin_serialize_variables(struct ast_channel *chan, char *buf, size_t size);
 extern char *pbx_builtin_getvar_helper(struct ast_channel *chan, const char *name);
+extern void pbx_builtin_pushvar_helper(struct ast_channel *chan, const char *name, const char *value);
 extern void pbx_builtin_setvar_helper(struct ast_channel *chan, const char *name, const char *value);
 extern void pbx_retrieve_variable(struct ast_channel *c, const char *var, char **ret, char *workspace, int workspacelen, struct varshead *headp);
 extern void pbx_builtin_clear_globals(void);

pbx.c

@@ -5897,6 +5897,26 @@ char *pbx_builtin_getvar_helper(struct ast_channel *chan, const char *name)
 	return NULL;
 }
 
+void pbx_builtin_pushvar_helper(struct ast_channel *chan, const char *name, const char *value)
+{
+	struct ast_var_t *newvariable;
+	struct varshead *headp;
+
+	if (name[strlen(name)-1] == ')') {
+		ast_log(LOG_WARNING, "Cannot push a value onto a function\n");
+		return ast_func_write(chan, name, value);
+	}
+
+	headp = (chan) ? &chan->varshead : &globals;
+
+	if (value) {
+		if ((option_verbose > 1) && (headp == &globals))
+			ast_verbose(VERBOSE_PREFIX_2 "Setting global variable '%s' to '%s'\n", name, value);
+		newvariable = ast_var_assign(name, value);	
+		AST_LIST_INSERT_HEAD(headp, newvariable, entries);
+	}
+}
+
 void pbx_builtin_setvar_helper(struct ast_channel *chan, const char *name, const char *value)
 {
 	struct ast_var_t *newvariable;