[asterisk-commits] branch rizzo/base r9688 - in /team/rizzo/base: ./ apps/ channels/ codecs/ doc...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Sun Feb 12 08:02:29 MST 2006 

Author: rizzo
Date: Sun Feb 12 09:02:06 2006
New Revision: 9688

URL: http://svn.digium.com/view/asterisk?rev=9688&view=rev
Log:
merge with current version and fix bugs.


Added:
    team/rizzo/base/channels/misdn_config.c
      - copied unchanged from r9674, trunk/channels/misdn_config.c
    team/rizzo/base/funcs/func_channel.c
      - copied unchanged from r9674, trunk/funcs/func_channel.c
Removed:
    team/rizzo/base/channels/chan_misdn_config.c
Modified:
    team/rizzo/base/   (props changed)
    team/rizzo/base/apps/app_curl.c
    team/rizzo/base/apps/app_hasnewvoicemail.c
    team/rizzo/base/apps/app_morsecode.c
    team/rizzo/base/apps/app_page.c
    team/rizzo/base/apps/app_queue.c
    team/rizzo/base/channel.c
    team/rizzo/base/channels/Makefile
    team/rizzo/base/channels/chan_agent.c
    team/rizzo/base/channels/chan_iax2.c
    team/rizzo/base/channels/chan_mgcp.c
    team/rizzo/base/channels/chan_sip.c
    team/rizzo/base/codecs/Makefile
    team/rizzo/base/doc/CODING-GUIDELINES
    team/rizzo/base/formats/format_g723.c
    team/rizzo/base/formats/format_ogg_vorbis.c
    team/rizzo/base/funcs/func_base64.c
    team/rizzo/base/funcs/func_callerid.c
    team/rizzo/base/funcs/func_cdr.c
    team/rizzo/base/funcs/func_cut.c
    team/rizzo/base/funcs/func_db.c
    team/rizzo/base/funcs/func_enum.c
    team/rizzo/base/funcs/func_env.c
    team/rizzo/base/funcs/func_groupcount.c
    team/rizzo/base/funcs/func_language.c
    team/rizzo/base/funcs/func_logic.c
    team/rizzo/base/funcs/func_math.c
    team/rizzo/base/funcs/func_md5.c
    team/rizzo/base/funcs/func_moh.c
    team/rizzo/base/funcs/func_odbc.c
    team/rizzo/base/funcs/func_rand.c
    team/rizzo/base/funcs/func_sha1.c
    team/rizzo/base/funcs/func_strings.c
    team/rizzo/base/funcs/func_timeout.c
    team/rizzo/base/funcs/func_uri.c
    team/rizzo/base/include/asterisk/channel.h
    team/rizzo/base/include/asterisk/manager.h
    team/rizzo/base/include/asterisk/module.h
    team/rizzo/base/include/asterisk/pbx.h
    team/rizzo/base/include/asterisk/stringfields.h
    team/rizzo/base/manager.c
    team/rizzo/base/pbx.c
    team/rizzo/base/pbx/pbx_dundi.c
    team/rizzo/base/pbx/pbx_spool.c
    team/rizzo/base/res/res_agi.c
    team/rizzo/base/res/res_clioriginate.c
    team/rizzo/base/utils.c

Propchange: team/rizzo/base/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.

Propchange: team/rizzo/base/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Propchange: team/rizzo/base/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Feb 12 09:02:06 2006
@@ -1,1 +1,1 @@
-/trunk:1-9580
+/trunk:1-9687

Modified: team/rizzo/base/apps/app_curl.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/apps/app_curl.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/apps/app_curl.c (original)
+++ team/rizzo/base/apps/app_curl.c Sun Feb 12 09:02:06 2006
@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C)  2004 - 2005, Tilghman Lesher
+ * Copyright (C)  2004 - 2006, Tilghman Lesher
  *
  * Tilghman Lesher <curl-20050919 at the-tilghman.com>
  * and Brian Wilkins <bwilkins at cfl.rr.com> (Added POST option)
@@ -109,10 +109,9 @@
 	return 0;
 }
 
-static char *acf_curl_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+static int acf_curl_exec(struct ast_channel *chan, char *cmd, char *info, char *buf, size_t len)
 {
 	struct localuser *u;
-	char *info;
 	struct MemoryStruct chunk = { NULL, 0 };
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(url);
@@ -121,21 +120,16 @@
 
 	*buf = '\0';
 	
-	if (ast_strlen_zero(data)) {
+	if (ast_strlen_zero(info)) {
 		ast_log(LOG_WARNING, "CURL requires an argument (URL)\n");
-		return buf;
+		return -1;
 	}
 
 	LOCAL_USER_ACF_ADD(u);
 
-	if (!(info = ast_strdupa(data))) {
-		LOCAL_USER_REMOVE(u);
-		return buf;
-	}
-
 	AST_STANDARD_APP_ARGS(args, info);	
 	
-	if (! curl_internal(&chunk, args.url, args.postdata)) {
+	if (!curl_internal(&chunk, args.url, args.postdata)) {
 		if (chunk.memory) {
 			chunk.memory[chunk.size] = '\0';
 			if (chunk.memory[chunk.size - 1] == 10)
@@ -149,7 +143,8 @@
 	}
 
 	LOCAL_USER_REMOVE(u);
-	return buf;
+
+	return 0;
 }
 
 struct ast_custom_function acf_curl = {

Modified: team/rizzo/base/apps/app_hasnewvoicemail.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/apps/app_hasnewvoicemail.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/apps/app_hasnewvoicemail.c (original)
+++ team/rizzo/base/apps/app_hasnewvoicemail.c Sun Feb 12 09:02:06 2006
@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Changes Copyright (c) 2004 - 2005 Todd Freeman <freeman at andrews.edu>
+ * Changes Copyright (c) 2004 - 2006 Todd Freeman <freeman at andrews.edu>
  * 
  * 95% based on HasNewVoicemail by:
  * 
@@ -178,10 +178,10 @@
 	return 0;
 }
 
-static char *acf_vmcount_exec(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+static int acf_vmcount_exec(struct ast_channel *chan, char *cmd, char *argsstr, char *buf, size_t len)
 {
 	struct localuser *u;
-	char *argsstr, *context;
+	char *context;
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(vmbox);
 		AST_APP_ARG(folder);
@@ -190,11 +190,6 @@
 	LOCAL_USER_ACF_ADD(u);
 
 	buf[0] = '\0';
-
-	if (!(argsstr = ast_strdupa(data))) {
-		LOCAL_USER_REMOVE(u);
-		return buf;
-	}
 
 	AST_STANDARD_APP_ARGS(args, argsstr);
 
@@ -213,7 +208,7 @@
 
 	LOCAL_USER_REMOVE(u);
 	
-	return buf;
+	return 0;
 }
 
 struct ast_custom_function acf_vmcount = {

Modified: team/rizzo/base/apps/app_morsecode.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/apps/app_morsecode.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/apps/app_morsecode.c (original)
+++ team/rizzo/base/apps/app_morsecode.c Sun Feb 12 09:02:06 2006
@@ -49,14 +49,14 @@
 
 static char *morsecode_descrip =
 "Usage: Morsecode(<string>)\n"
-"Plays the Morse code equivalent of the passed string\n";
+"Plays the Morse code equivalent of the passed string.  If the variable\n"
+"MORSEDITLEN is set, it will use that value for the length (in ms) of the dit\n"
+"(defaults to 80).  Additionally, if MORSETONE is set, it will use that tone\n"
+"(in Hz).  The tone default is 800.\n";
 
 STANDARD_LOCAL_USER;
 
 LOCAL_USER_DECL;
-
-#define	TONE	800
-#define	DITLEN	80
 
 static char *morsecode[] = {
 	"", "", "", "", "", "", "", "", "", "", "", "", "", "", "", "", /*  0-15 */
@@ -105,16 +105,17 @@
 static void playtone(struct ast_channel *chan, int tone, int len)
 {
 	char dtmf[20];
-	snprintf(dtmf, sizeof(dtmf), "%d/%d", tone, DITLEN * len);
+	snprintf(dtmf, sizeof(dtmf), "%d/%d", tone, len);
 	ast_playtones_start(chan, 0, dtmf, 0);
-	ast_safe_sleep(chan, DITLEN * len);
+	ast_safe_sleep(chan, len);
 	ast_playtones_stop(chan);
 }
 
 static int morsecode_exec(struct ast_channel *chan, void *data)
 {
-	int res=0;
+	int res=0, ditlen, tone;
 	char *digit;
+	const char *ditlenc, *tonec;
 	struct localuser *u;
 
 	LOCAL_USER_ADD(u);
@@ -125,6 +126,18 @@
 		return 0;
 	}
 
+	/* Use variable MORESEDITLEN, if set (else 80) */
+	ditlenc = pbx_builtin_getvar_helper(chan, "MORSEDITLEN");
+	if (ast_strlen_zero(ditlenc) || (sscanf(ditlenc, "%d", &ditlen) != 1)) {
+		ditlen = 80;
+	}
+
+	/* Use variable MORSETONE, if set (else 800) */
+	tonec = pbx_builtin_getvar_helper(chan, "MORSETONE");
+	if (ast_strlen_zero(tonec) || (sscanf(tonec, "%d", &tone) != 1)) {
+		tone = 800;
+	}
+
 	for (digit = data; *digit; digit++) {
 		char *dahdit;
 		if (*digit < 0) {
@@ -132,19 +145,19 @@
 		}
 		for (dahdit = morsecode[(int)*digit]; *dahdit; dahdit++) {
 			if (*dahdit == '-') {
-				playtone(chan, TONE, 3);
+				playtone(chan, tone, 3 * ditlen);
 			} else if (*dahdit == '.') {
-				playtone(chan, TONE, 1);
+				playtone(chan, tone, 1 * ditlen);
 			} else {
 				/* Account for ditlen of silence immediately following */
-				playtone(chan, 0, 2);
+				playtone(chan, 0, 2 * ditlen);
 			}
 
 			/* Pause slightly between each dit and dah */
-			playtone(chan, 0, 1);
+			playtone(chan, 0, 1 * ditlen);
 		}
 		/* Pause between characters */
-		playtone(chan, 0, 2);
+		playtone(chan, 0, 2 * ditlen);
 	}
 
 	LOCAL_USER_REMOVE(u);

Modified: team/rizzo/base/apps/app_page.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/apps/app_page.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/apps/app_page.c (original)
+++ team/rizzo/base/apps/app_page.c Sun Feb 12 09:02:06 2006
@@ -87,7 +87,7 @@
 {
 	struct calloutdata *cd = data;
 	ast_pbx_outgoing_app(cd->tech, AST_FORMAT_SLINEAR, cd->resource, 30000,
-		"MeetMe", cd->meetmeopts, NULL, 0, cd->cidnum, cd->cidname, cd->variables, NULL);
+		"MeetMe", cd->meetmeopts, NULL, 0, cd->cidnum, cd->cidname, cd->variables, NULL, NULL);
 	free(cd);
 	return NULL;
 }

Modified: team/rizzo/base/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/apps/app_queue.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/apps/app_queue.c (original)
+++ team/rizzo/base/apps/app_queue.c Sun Feb 12 09:02:06 2006
@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Copyright (C) 1999 - 2006, Digium, Inc.
  *
  * Mark Spencer <markster at digium.com>
  *
@@ -3093,19 +3093,18 @@
 	return res;
 }
 
-static char *queue_function_qac(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+static int queue_function_qac(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
 {
 	int count = 0;
 	struct ast_call_queue *q;
 	struct localuser *lu;
 	struct member *m;
 
-
-	ast_copy_string(buf, "0", len);
+	buf[0] = '\0';
 	
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_ERROR, "%s requires an argument: queuename\n", cmd);
-		return buf;
+		return -1;
 	}
 
 	LOCAL_USER_ACF_ADD(lu);
@@ -3134,10 +3133,10 @@
 
 	snprintf(buf, len, "%d", count);
 	LOCAL_USER_REMOVE(lu);
-	return buf;
-}
-
-static char *queue_function_queuememberlist(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+	return 0;
+}
+
+static int queue_function_queuememberlist(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
 {
 	struct localuser *u;
 	struct ast_call_queue *q;
@@ -3148,7 +3147,7 @@
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_ERROR, "QUEUE_MEMBER_LIST requires an argument: queuename\n");
-		return buf;
+		return -1;
 	}
 	
 	LOCAL_USER_ACF_ADD(u);
@@ -3187,7 +3186,7 @@
 	/* We should already be terminated, but let's make sure. */
 	buf[len - 1] = '\0';
 	LOCAL_USER_REMOVE(u);
-	return buf;
+	return 0;
 }
 
 static struct ast_custom_function queueagentcount_function = {

Modified: team/rizzo/base/channel.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/channel.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/channel.c (original)
+++ team/rizzo/base/channel.c Sun Feb 12 09:02:06 2006
@@ -2421,6 +2421,8 @@
 				ast_set_callerid(chan, oh->cid_num, oh->cid_name, oh->cid_num);
 			if (oh->parent_channel)
 				ast_channel_inherit_variables(oh->parent_channel, chan);
+			if (oh->account)
+				ast_cdr_setaccount(chan, oh->account);	
 		}
 		ast_set_callerid(chan, cid_num, cid_name, cid_num);
 

Modified: team/rizzo/base/channels/Makefile
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/channels/Makefile?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/channels/Makefile (original)
+++ team/rizzo/base/channels/Makefile Sun Feb 12 09:02:06 2006
@@ -235,15 +235,14 @@
 misdn/chan_misdn_lib.a:
 	make -C misdn
 
-chan_misdn.so: chan_misdn.o chan_misdn_config.o misdn/chan_misdn_lib.a
+chan_misdn.so: chan_misdn.o misdn_config.o misdn/chan_misdn_lib.a
 	$(CC) -shared -Xlinker -x -L/usr/lib -o $@ $^ -lisdnnet -lmISDN
 
 chan_misdn.o: chan_misdn.c
 	$(CC) $(CFLAGS) -DCHAN_MISDN_VERSION=\"0.3.0\" -c $< -o $@
 
-chan_misdn_config.o: chan_misdn_config.c misdn/chan_misdn_config.h
+misdn_config.o: misdn_config.c misdn/chan_misdn_config.h
 	$(CC) $(CFLAGS) -DCHAN_MISDN_VERSION=\"0.3.0\" -c $< -o $@
-
 
 install: all
 	for x in $(MODS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done

Modified: team/rizzo/base/channels/chan_agent.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/channels/chan_agent.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/channels/chan_agent.c (original)
+++ team/rizzo/base/channels/chan_agent.c Sun Feb 12 09:02:06 2006
@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Copyright (C) 1999 - 2006, Digium, Inc.
  *
  * Mark Spencer <markster at digium.com>
  *
@@ -2406,7 +2406,7 @@
 	return cur;	
 }
 
-static char *function_agent(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+static int function_agent(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
 {
 	char *parse;    
 	AST_DECLARE_APP_ARGS(args,
@@ -2420,11 +2420,11 @@
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "The AGENT function requires an argument - agentid!\n");
-		return buf;	
+		return -1;
 	}
 
 	if (!(parse = ast_strdupa(data)))
-		return buf;
+		return -1;
 
 	AST_NONSTANDARD_APP_ARGS(args, parse, ':');
 	if (!args.item)
@@ -2432,7 +2432,7 @@
 
 	if (!(agent = find_agent(args.agentid))) {
 		ast_log(LOG_WARNING, "Agent '%s' not found!\n", args.agentid);
-		return buf;
+		return -1;
 	}
 
 	if (!strcasecmp(args.item, "status")) {
@@ -2458,7 +2458,7 @@
 		ast_copy_string(buf, agent->loginchan, len);	
 	}
 
-	return buf;
+	return 0;
 }
 
 struct ast_custom_function agent_function = {

Modified: team/rizzo/base/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/channels/chan_iax2.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/channels/chan_iax2.c (original)
+++ team/rizzo/base/channels/chan_iax2.c Sun Feb 12 09:02:06 2006
@@ -9145,31 +9145,29 @@
 	return -1;
 }
 
-static char *function_iaxpeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
-{
-	char *ret = NULL;
+static int function_iaxpeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
 	struct iax2_peer *peer;
 	char *peername, *colname;
 	char iabuf[INET_ADDRSTRLEN];
 
 	if (!(peername = ast_strdupa(data)))
-		return ret;
+		return -1;
 
 	/* if our channel, return the IP address of the endpoint of current channel */
 	if (!strcmp(peername,"CURRENTCHANNEL")) {
 	        unsigned short callno = PTR_TO_CALLNO(chan->tech_pvt);
 		ast_copy_string(buf, iaxs[callno]->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iaxs[callno]->addr.sin_addr) : "", len);
-		return buf;
-	}
-
-	if ((colname = strchr(peername, ':'))) {
-		*colname = '\0';
-		colname++;
-	} else {
+		return 0;
+	}
+
+	if ((colname = strchr(peername, ':')))
+		*colname++ = '\0';
+	else
 		colname = "ip";
-	}
+
 	if (!(peer = find_peer(peername, 1)))
-		return ret;
+		return -1;
 
 	if (!strcasecmp(colname, "ip")) {
 		ast_copy_string(buf, peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "", len);
@@ -9204,16 +9202,15 @@
 			ast_copy_string(buf, ast_getformatname(codec), len);
 		}
 	}
-	ret = buf;
-
-	return ret;
+
+	return 0;
 }
 
 struct ast_custom_function iaxpeer_function = {
-    .name = "IAXPEER",
-    .synopsis = "Gets IAX peer information",
-    .syntax = "IAXPEER(<peername|CURRENTCHANNEL>[:item])",
-    .read = function_iaxpeer,
+	.name = "IAXPEER",
+	.synopsis = "Gets IAX peer information",
+	.syntax = "IAXPEER(<peername|CURRENTCHANNEL>[:item])",
+	.read = function_iaxpeer,
 	.desc = "If peername specified, valid items are:\n"
 	"- ip (default)          The IP address.\n"
 	"- status                The peer's status (if qualify=yes)\n"
@@ -9459,6 +9456,7 @@
 	ast_channel_unregister(&iax2_tech);
 	delete_users();
 	iax_provision_unload();
+	sched_context_destroy(sched);
 	return 0;
 }
 

Modified: team/rizzo/base/channels/chan_mgcp.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/channels/chan_mgcp.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/channels/chan_mgcp.c (original)
+++ team/rizzo/base/channels/chan_mgcp.c Sun Feb 12 09:02:06 2006
@@ -4450,6 +4450,7 @@
 	ast_cli_unregister(&cli_debug);
 	ast_cli_unregister(&cli_no_debug);
 	ast_cli_unregister(&cli_mgcp_reload);
+	sched_context_destroy(sched);
 
 	return 0;
 }

Modified: team/rizzo/base/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/channels/chan_sip.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/channels/chan_sip.c (original)
+++ team/rizzo/base/channels/chan_sip.c Sun Feb 12 09:02:06 2006
@@ -9021,21 +9021,21 @@
 
 
 /*! \brief  func_header_read: Read SIP header (dialplan function) */
-static char *func_header_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) 
+int func_header_read(struct ast_channel *chan, char *function, char *data, char *buf, size_t len) 
 {
 	struct sip_pvt *p;
 	const char *content;
 	
  	if (!data) {
 		ast_log(LOG_WARNING, "This function requires a header name.\n");
-		return NULL;
+		return -1;
 	}
 
 	ast_mutex_lock(&chan->lock);
 	if (chan->tech != &sip_tech) {
 		ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
 		ast_mutex_unlock(&chan->lock);
-		return NULL;
+		return -1;
 	}
 
 	p = chan->tech_pvt;
@@ -9043,22 +9043,21 @@
 	/* If there is no private structure, this channel is no longer alive */
 	if (!p) {
 		ast_mutex_unlock(&chan->lock);
-		return NULL;
+		return -1;
 	}
 
 	content = get_header(&p->initreq, data);
 
 	if (ast_strlen_zero(content)) {
 		ast_mutex_unlock(&chan->lock);
-		return NULL;
+		return -1;
 	}
 
 	ast_copy_string(buf, content, len);
 	ast_mutex_unlock(&chan->lock);
 
-	return buf;
-}
-
+	return 0;
+}
 
 static struct ast_custom_function sip_header_function = {
 	.name = "SIP_HEADER",
@@ -9068,17 +9067,17 @@
 };
 
 /*! \brief  function_check_sipdomain: Dial plan function to check if domain is local */
-static char *func_check_sipdomain(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+int func_check_sipdomain(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
 {
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "CHECKSIPDOMAIN requires an argument - A domain name\n");
-		return buf;
+		return -1;
 	}
 	if (check_sip_domain(data, NULL, 0))
 		ast_copy_string(buf, data, len);
 	else
 		buf[0] = '\0';
-	return buf;
+	return 0;
 }
 
 static struct ast_custom_function checksipdomain_function = {
@@ -9092,26 +9091,20 @@
 		"Check the domain= configuration in sip.conf\n",
 };
 
-
 /*! \brief  function_sippeer: ${SIPPEER()} Dialplan function - reads peer data */
-static char *function_sippeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
-{
-	char *ret = NULL;
+static int function_sippeer(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len)
+{
 	struct sip_peer *peer;
-	char *peername, *colname;
+	char *colname;
 	char iabuf[INET_ADDRSTRLEN];
 
-	if (!(peername = ast_strdupa(data)))
-		return ret;
-
-	if ((colname = strchr(peername, ':'))) {
-		*colname = '\0';
-		colname++;
-	} else {
+	if ((colname = strchr(data, ':')))
+		*colname++ = '\0';
+	else
 		colname = "ip";
-	}
-	if (!(peer = find_peer(peername, NULL, 1)))
-		return ret;
+
+	if (!(peer = find_peer(data, NULL, 1)))
+		return -1;
 
 	if (!strcasecmp(colname, "ip")) {
 		ast_copy_string(buf, peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "", len);
@@ -9148,19 +9141,18 @@
 		codecnum = strchr(colname, '[');
 		*codecnum = '\0';
 		codecnum++;
-		if ((ptr = strchr(codecnum, ']'))) {
+		if ((ptr = strchr(codecnum, ']')))
 			*ptr = '\0';
-		}
+
 		index = atoi(codecnum);
 		if((codec = ast_codec_pref_index(&peer->prefs, index))) {
 			ast_copy_string(buf, ast_getformatname(codec), len);
 		}
 	}
-	ret = buf;
 
 	ASTOBJ_UNREF(peer, sip_destroy_peer);
 
-	return ret;
+	return 0;
 }
 
 /*! \brief Structure to declare a dialplan function: SIPPEER */
@@ -9190,7 +9182,7 @@
 };
 
 /*! \brief  function_sipchaninfo_read: ${SIPCHANINFO()} Dialplan function - reads sip channel data */
-static char *function_sipchaninfo_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) 
+int function_sipchaninfo_read(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) 
 {
 	struct sip_pvt *p;
 	char iabuf[INET_ADDRSTRLEN];
@@ -9199,14 +9191,14 @@
 	
  	if (!data) {
 		ast_log(LOG_WARNING, "This function requires a parameter name.\n");
-		return NULL;
+		return -1;
 	}
 
 	ast_mutex_lock(&chan->lock);
 	if (chan->tech != &sip_tech) {
 		ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
 		ast_mutex_unlock(&chan->lock);
-		return NULL;
+		return -1;
 	}
 
 /* 	ast_verbose("function_sipchaninfo_read: %s\n", data); */
@@ -9215,7 +9207,7 @@
 	/* If there is no private structure, this channel is no longer alive */
 	if (!p) {
 		ast_mutex_unlock(&chan->lock);
-		return NULL;
+		return -1;
 	}
 
 	if (!strcasecmp(data, "peerip")) {
@@ -9232,11 +9224,11 @@
 		ast_copy_string(buf, p->peername, len);
 	} else {
 		ast_mutex_unlock(&chan->lock);
-		return NULL;
+		return -1;
 	}
 	ast_mutex_unlock(&chan->lock);
 
-	return buf;
+	return 0;
 }
 
 /*! \brief Structure to declare a dialplan function: SIPCHANINFO */
@@ -9253,8 +9245,6 @@
 	"- useragent             The useragent.\n"
 	"- peername              The name of the peer.\n"
 };
-
-
 
 /*! \brief  parse_moved_contact: Parse 302 Moved temporalily response */
 static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
@@ -13003,6 +12993,7 @@
 	clear_realm_authentication(authl);
 	clear_sip_domains();
 	close(sipsock);
+	sched_context_destroy(sched);
 		
 	return 0;
 }

Modified: team/rizzo/base/codecs/Makefile
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/codecs/Makefile?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/codecs/Makefile (original)
+++ team/rizzo/base/codecs/Makefile Sun Feb 12 09:02:06 2006
@@ -33,7 +33,7 @@
 endif
 
 SPEEX_PATH:=/usr/local/include /usr/include /usr/include/speex /usr/local/include/speex
-SPEEX_SYSTEM_HEADERS:=$(wildcard $(SPEEX_PATH:%=$(CROSS_COMPILE_TARGET)%/speex.h))
+SPEEX_SYSTEM_HEADERS:=$(firstword $(wildcard $(SPEEX_PATH:%=$(CROSS_COMPILE_TARGET)%/speex.h)))
 ifeq (${SPEEX_SYSTEM_HEADERS},)
   MODS:=$(filter-out codec_speex.so,$(MODS))
 else
@@ -48,7 +48,7 @@
 endif
 
 LIBGSM_PATH:=/usr/local/include /usr/include
-LIBGSM_SYSTEM_HEADERS:=$(wildcard $(LIBGSM_PATH:%=$(CROSS_COMPILE_TARGET)%/gsm/gsm.h))
+LIBGSM_SYSTEM_HEADERS:=$(firstword $(wildcard $(LIBGSM_PATH:%=$(CROSS_COMPILE_TARGET)%/gsm/gsm.h)))
 ifneq ($(LIBGSM_SYSTEM_HEADERS),)
   LIBGSM=-lgsm
   LIBGSMT=

Modified: team/rizzo/base/doc/CODING-GUIDELINES
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/doc/CODING-GUIDELINES?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/doc/CODING-GUIDELINES (original)
+++ team/rizzo/base/doc/CODING-GUIDELINES Sun Feb 12 09:02:06 2006
@@ -77,7 +77,7 @@
 Roughly, Asterisk code formatting guidelines are generally equivalent to the 
 following:
 
-# indent -i4 -ts4 -br -brs -cdw -cli0 -ce -nbfda -npcs -nprs -npsl -saf -sai -saw foo.c
+# indent -i4 -ts4 -br -brs -cdw -lp -ce -nbfda -npcs -nprs -npsl -nbbo -saf -sai -saw -cs -ln90 foo.c
 
 this means in verbose:
  -i4:    indent level 4
@@ -85,7 +85,7 @@
  -br:    braces on if line
  -brs:   braces on struct decl line
  -cdw:   cuddle do while
- -cli0:  case indentation 0
+ -lp:    line up continuation below parenthesis
  -ce:    cuddle else
  -nbfda: dont break function decl args
  -npcs:  no space after function call names
@@ -94,6 +94,8 @@
  -saf:   space after for
  -sai:   space after if
  -saw:   space after while
+ -cs:    space after cast
+ -ln90:  line length 90 columns
 
 Function calls and arguments should be spaced in a consistent way across
 the codebase.

Modified: team/rizzo/base/formats/format_g723.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/formats/format_g723.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/formats/format_g723.c (original)
+++ team/rizzo/base/formats/format_g723.c Sun Feb 12 09:02:06 2006
@@ -44,9 +44,6 @@
 #include "asterisk/logger.h"
 #include "asterisk/sched.h"
 #include "asterisk/module.h"
-
-#include "../channels/adtranvofr.h"
-
 
 #define G723_MAX_SIZE 1024
 

Modified: team/rizzo/base/formats/format_ogg_vorbis.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/formats/format_ogg_vorbis.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/formats/format_ogg_vorbis.c (original)
+++ team/rizzo/base/formats/format_ogg_vorbis.c Sun Feb 12 09:02:06 2006
@@ -20,7 +20,7 @@
  * \arg File name extension: ogg
  * \ingroup formats
  */
- 
+
 #include <sys/types.h>
 #include <netinet/in.h>
 #include <arpa/inet.h>
@@ -48,32 +48,31 @@
 #include "asterisk/file.h"
 #include "asterisk/logger.h"
 #include "asterisk/module.h"
-
 #define SAMPLES_MAX 160
 #define BLOCK_SIZE 4096
 
-
 struct vorbis_desc {
 	/* structures for handling the Ogg container */
-	ogg_sync_state	 oy;
+	ogg_sync_state oy;
 	ogg_stream_state os;
-	ogg_page	 og;
-	ogg_packet	 op;
+	ogg_page og;
+	ogg_packet op;
 	
 	/* structures for handling Vorbis audio data */
-	vorbis_info	 vi;
-	vorbis_comment	 vc;
+	vorbis_info vi;
+	vorbis_comment vc;
 	vorbis_dsp_state vd;
-	vorbis_block	 vb;
+	vorbis_block vb;
 	
 	/*! \brief Indicates whether this filestream is set up for reading or writing. */
 	int writing;
-
+	
 	/*! \brief Indicates whether an End of Stream condition has been detected. */
 	int eos;
 };
 
 AST_MUTEX_DEFINE_STATIC(ogg_vorbis_lock);
+
 static int glistcnt = 0;
 
 static char *name = "ogg_vorbis";
@@ -210,7 +209,8 @@
  * \param comment Comment that should be embedded in the OGG/Vorbis file.
  * \return A new filestream.
  */
-static struct ast_filestream *ogg_vorbis_rewrite(FILE *f, const char *comment)
+static struct ast_filestream *ogg_vorbis_rewrite(FILE * f,
+						 const char *comment)
 {
 	ogg_packet header;
 	ogg_packet header_comm;
@@ -218,7 +218,7 @@
 
 	struct ast_filestream *tmp;
 
-	if((tmp = malloc(sizeof(struct ast_filestream)))) {
+	if ((tmp = malloc(sizeof(struct ast_filestream)))) {
 		memset(tmp, 0, sizeof(struct ast_filestream));
 
 		tmp->writing = 1;
@@ -226,7 +226,7 @@
 
 		vorbis_info_init(&tmp->vi);
 
-		if(vorbis_encode_init_vbr(&tmp->vi, 1, 8000, 0.4)) {
+		if (vorbis_encode_init_vbr(&tmp->vi, 1, 8000, 0.4)) {
 			ast_log(LOG_ERROR, "Unable to initialize Vorbis encoder!\n");
 			free(tmp);
 			return NULL;
@@ -234,7 +234,7 @@
 
 		vorbis_comment_init(&tmp->vc);
 		vorbis_comment_add_tag(&tmp->vc, "ENCODER", "Asterisk PBX");
-		if(comment)
+		if (comment)
 			vorbis_comment_add_tag(&tmp->vc, "COMMENT", (char *) comment);
 
 		vorbis_analysis_init(&tmp->vd, &tmp->vi);
@@ -242,21 +242,22 @@
 
 		ogg_stream_init(&tmp->os, rand());
 
-		vorbis_analysis_headerout(&tmp->vd, &tmp->vc, &header, &header_comm, &header_code);
-		ogg_stream_packetin(&tmp->os, &header);							
+		vorbis_analysis_headerout(&tmp->vd, &tmp->vc, &header, &header_comm,
+					  &header_code);
+		ogg_stream_packetin(&tmp->os, &header);
 		ogg_stream_packetin(&tmp->os, &header_comm);
 		ogg_stream_packetin(&tmp->os, &header_code);
 
-		while(!tmp->eos) {
-			if(ogg_stream_flush(&tmp->os, &tmp->og) == 0)
+		while (!tmp->eos) {
+			if (ogg_stream_flush(&tmp->os, &tmp->og) == 0)
 				break;
 			fwrite(tmp->og.header, 1, tmp->og.header_len, tmp->f);
 			fwrite(tmp->og.body, 1, tmp->og.body_len, tmp->f);
-			if(ogg_page_eos(&tmp->og))
+			if (ogg_page_eos(&tmp->og))
 				tmp->eos = 1;
 		}
 
-		if(ast_mutex_lock(&ogg_vorbis_lock)) {
+		if (ast_mutex_lock(&ogg_vorbis_lock)) {
 			ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
 			fclose(f);
 			ogg_stream_clear(&tmp->os);
@@ -283,16 +284,16 @@
 	while (vorbis_analysis_blockout(&s->vd, &s->vb) == 1) {
 		vorbis_analysis(&s->vb, NULL);
 		vorbis_bitrate_addblock(&s->vb);
-		
+
 		while (vorbis_bitrate_flushpacket(&s->vd, &s->op)) {
 			ogg_stream_packetin(&s->os, &s->op);
 			while (!s->eos) {
-				if(ogg_stream_pageout(&s->os, &s->og) == 0) {
+				if (ogg_stream_pageout(&s->os, &s->og) == 0) {
 					break;
 				}
 				fwrite(s->og.header, 1, s->og.header_len, s->f);
 				fwrite(s->og.body, 1, s->og.body_len, s->f);
-				if(ogg_page_eos(&s->og)) {
+				if (ogg_page_eos(&s->og)) {
 					s->eos = 1;
 				}
 			}
@@ -312,20 +313,21 @@
 	float **buffer;
 	short *data;
 
-	if(!s->writing) {
+	if (!s->writing) {
 		ast_log(LOG_ERROR, "This stream is not set up for writing!\n");
 		return -1;
 	}
 
-	if(f->frametype != AST_FRAME_VOICE) {
+	if (f->frametype != AST_FRAME_VOICE) {
 		ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
 		return -1;
 	}
-	if(f->subclass != AST_FORMAT_SLINEAR) {
-		ast_log(LOG_WARNING, "Asked to write non-SLINEAR frame (%d)!\n", f->subclass);
+	if (f->subclass != AST_FORMAT_SLINEAR) {
+		ast_log(LOG_WARNING, "Asked to write non-SLINEAR frame (%d)!\n",
+				f->subclass);
 		return -1;
 	}
-	if(!f->datalen)
+	if (!f->datalen)
 		return -1;
 
 	data = (short *) f->data;
@@ -333,7 +335,7 @@
 	buffer = vorbis_analysis_buffer(&s->vd, f->samples);
 
 	for (i = 0; i < f->samples; i++) {
-		buffer[0][i] = data[i]/32768.f;
+		buffer[0][i] = data[i] / 32768.f;
 	}
 
 	vorbis_analysis_wrote(&s->vd, f->samples);
@@ -349,7 +351,7 @@
  */
 static void ogg_vorbis_close(struct ast_filestream *s)
 {
-	if(ast_mutex_lock(&ogg_vorbis_lock)) {
+	if (ast_mutex_lock(&ogg_vorbis_lock)) {
 		ast_log(LOG_WARNING, "Unable to lock ogg_vorbis list\n");
 		return;
 	}
@@ -357,7 +359,7 @@
 	ast_mutex_unlock(&ogg_vorbis_lock);
 	ast_update_use_count();
 
-	if(s->writing) {
+	if (s->writing) {
 		/* Tell the Vorbis encoder that the stream is finished
 		 * and write out the rest of the data */
 		vorbis_analysis_wrote(&s->vd, 0);
@@ -370,7 +372,7 @@
 	vorbis_comment_clear(&s->vc);
 	vorbis_info_clear(&s->vi);
 
-	if(s->writing) {
+	if (s->writing) {
 		ogg_sync_clear(&s->oy);
 	}
 }
@@ -390,28 +392,29 @@
 
 	while (1) {
 		samples_in = vorbis_synthesis_pcmout(&s->vd, pcm);
-		if(samples_in > 0) {
+		if (samples_in > 0) {
 			return samples_in;
 		}
-		
+
 		/* The Vorbis decoder needs more data... */
 		/* See ifOGG has any packets in the current page for the Vorbis decoder. */
 		result = ogg_stream_packetout(&s->os, &s->op);
-		if(result > 0) {
+		if (result > 0) {
 			/* Yes OGG had some more packets for the Vorbis decoder. */
-			if(vorbis_synthesis(&s->vb, &s->op) == 0) {
+			if (vorbis_synthesis(&s->vb, &s->op) == 0) {
 				vorbis_synthesis_blockin(&s->vd, &s->vb);
 			}
-			
+
 			continue;
 		}
 
-		if(result < 0)
-			ast_log(LOG_WARNING, "Corrupt or missing data at this page position; continuing...\n");
-		
+		if (result < 0)
+			ast_log(LOG_WARNING,
+					"Corrupt or missing data at this page position; continuing...\n");
+
 		/* No more packets left in the current page... */
 
-		if(s->eos) {
+		if (s->eos) {
 			/* No more pages left in the stream */
 			return -1;
 		}
@@ -419,22 +422,24 @@
 		while (!s->eos) {
 			/* See ifOGG has any pages in it's internal buffers */
 			result = ogg_sync_pageout(&s->oy, &s->og);
-			if(result > 0) {
+			if (result > 0) {
 				/* Yes, OGG has more pages in it's internal buffers,
 				   add the page to the stream state */
 				result = ogg_stream_pagein(&s->os, &s->og);
-				if(result == 0) {
+				if (result == 0) {
 					/* Yes, got a new,valid page */
-					if(ogg_page_eos(&s->og)) {
+					if (ogg_page_eos(&s->og)) {
 						s->eos = 1;
 					}
 					break;
 				}
-				ast_log(LOG_WARNING, "Invalid page in the bitstream; continuing...\n");
-			}
-			
-			if(result < 0)
-				ast_log(LOG_WARNING, "Corrupt or missing data in bitstream; continuing...\n");
+				ast_log(LOG_WARNING,
+						"Invalid page in the bitstream; continuing...\n");
+			}
+
+			if (result < 0)
+				ast_log(LOG_WARNING,
+						"Corrupt or missing data in bitstream; continuing...\n");
 
 			/* No, we need to read more data from the file descrptor */
 			/* get a buffer from OGG to read the data into */
@@ -443,7 +448,7 @@
 			bytes = fread(buffer, 1, BLOCK_SIZE, s->f);
 			/* Tell OGG how many bytes we actually read into the buffer */
 			ogg_sync_wrote(&s->oy, bytes);
-			if(bytes == 0) {
+			if (bytes == 0) {
 				s->eos = 1;
 			}
 		}
@@ -456,7 +461,8 @@
  * \param whennext Number of sample times to schedule the next call.
  * \return A pointer to a frame containing audio data or NULL ifthere is no more audio data.
  */
-static struct ast_frame *ogg_vorbis_read(struct ast_filestream *s, int *whennext)
+static struct ast_frame *ogg_vorbis_read(struct ast_filestream *s,
+					 int *whennext)
 {
 	int clipflag = 0;
 	int i;
@@ -470,25 +476,25 @@
 
 	while (1) {
 		/* See ifwe have filled up an audio frame yet */
-		if(samples_out == SAMPLES_MAX)
+		if (samples_out == SAMPLES_MAX)
 			break;
 
 		/* See ifVorbis decoder has some audio data for us ... */
 		samples_in = read_samples(s, &pcm);
-		if(samples_in <= 0)
+		if (samples_in <= 0)
 			break;
 
 		/* Got some audio data from Vorbis... */
 		/* Convert the float audio data to 16-bit signed linear */
-		
+
 		clipflag = 0;
 
 		samples_in = samples_in < (SAMPLES_MAX - samples_out) ? samples_in : (SAMPLES_MAX - samples_out);
-  
-		for(j = 0; j < samples_in; j++)
+
+		for (j = 0; j < samples_in; j++)
 			accumulator[j] = 0.0;
 
-		for(i = 0; i < s->vi.channels; i++) {
+		for (i = 0; i < s->vi.channels; i++) {
 			mono = pcm[i];
 			for (j = 0; j < samples_in; j++) {
 				accumulator[j] += mono[j];
@@ -496,27 +502,26 @@
 		}
 
 		for (j = 0; j < samples_in; j++) {
-			val =  accumulator[j] * 32767.0 / s->vi.channels;
-			if(val > 32767) {
+			val = accumulator[j] * 32767.0 / s->vi.channels;
+			if (val > 32767) {
 				val = 32767;
 				clipflag = 1;
 			}
-			if(val < -32768) {
+			if (val < -32768) {
 				val = -32768;
 				clipflag = 1;
 			}
 			s->buffer[samples_out + j] = val;
 		}
-			
-		if(clipflag)
-			ast_log(LOG_WARNING, "Clipping in frame %ld\n", (long)(s->vd.sequence));
-		
+
+		if (clipflag)
+			ast_log(LOG_WARNING, "Clipping in frame %ld\n", (long) (s->vd.sequence));
 		/* Tell the Vorbis decoder how many samples we actually used. */
 		vorbis_synthesis_read(&s->vd, samples_in);
 		samples_out += samples_in;
 	}
 
-	if(samples_out > 0) {
+	if (samples_out > 0) {
 		s->fr.frametype = AST_FRAME_VOICE;
 		s->fr.subclass = AST_FORMAT_SLINEAR;
 		s->fr.offset = AST_FRIENDLY_OFFSET;
@@ -526,7 +531,7 @@
 		s->fr.mallocd = 0;
 		s->fr.samples = samples_out;
 		*whennext = samples_out;
-		
+
 		return &s->fr;
 	} else {
 		return NULL;
@@ -553,17 +558,21 @@
  * \return 0 on success, -1 on failure.
  */
 
-static int ogg_vorbis_seek(struct ast_filestream *s, long sample_offset, int whence) {
+static int ogg_vorbis_seek(struct ast_filestream *s, long sample_offset,
+			   int whence)
+{
 	ast_log(LOG_WARNING, "Seeking is not supported on OGG/Vorbis streams!\n");
 	return -1;
 }
 
-static long ogg_vorbis_tell(struct ast_filestream *s) {
+static long ogg_vorbis_tell(struct ast_filestream *s)
+{
 	ast_log(LOG_WARNING, "Telling is not supported on OGG/Vorbis streams!\n");
 	return -1;
 }
 
-static char *ogg_vorbis_getcomment(struct ast_filestream *s) {
+static char *ogg_vorbis_getcomment(struct ast_filestream *s)
+{
 	ast_log(LOG_WARNING, "Getting comments is not supported on OGG/Vorbis streams!\n");
 	return NULL;
 }
@@ -595,7 +604,7 @@
 int unload_module()
 {
 	return ast_format_unregister(name);
-}	
+}
 
 int usecount()
 {
@@ -612,11 +621,3 @@
 {
 	return ASTERISK_GPL_KEY;
 }
-
-/*
-Local Variables:
-mode: C
-c-file-style: "linux"
-indent-tabs-mode: t
-End:
-*/

Modified: team/rizzo/base/funcs/func_base64.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/funcs/func_base64.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/funcs/func_base64.c (original)
+++ team/rizzo/base/funcs/func_base64.c Sun Feb 12 09:02:06 2006
@@ -36,31 +36,30 @@
 #include "asterisk/utils.h"
 #include "asterisk/app.h"
 
-static char *base64_encode(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) 
+static int base64_encode(struct ast_channel *chan, char *cmd, char *data,
+			 char *buf, size_t len)
 {
-	int res = 0;
-
-	if (ast_strlen_zero(data) ) {
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Syntax: BASE64_ENCODE(<data>) - missing argument!\n");
-		return NULL;
+		return -1;
 	}
 
-	ast_log(LOG_DEBUG, "data=%s\n",data);
-	res = ast_base64encode(buf, (unsigned char *)data, strlen(data), len);
-	ast_log(LOG_DEBUG, "res=%d\n", res);
-	return buf;
+	ast_base64encode(buf, (unsigned char *) data, strlen(data), len);
+
+	return 0;
 }
 
-static char *base64_decode(struct ast_channel *chan, char *cmd, char *data, char *buf, size_t len) 
+static int base64_decode(struct ast_channel *chan, char *cmd, char *data,
+			 char *buf, size_t len)
 {
-	if (ast_strlen_zero(data) ) {
+	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "Syntax: BASE64_DECODE(<base_64 string>) - missing argument!\n");
-		return NULL;
+		return -1;
 	}
 
-	ast_log(LOG_DEBUG, "data=%s\n", data);
-	ast_base64decode((unsigned char *)buf, data, len);
-	return buf;
+	ast_base64decode((unsigned char *) buf, data, len);
+
+	return 0;
 }
 
 static struct ast_custom_function base64_encode_function = {
@@ -83,13 +82,13 @@
 
 int unload_module(void)
 {
-        return ast_custom_function_unregister(&base64_encode_function) ||
+	return ast_custom_function_unregister(&base64_encode_function) |
 		ast_custom_function_unregister(&base64_decode_function);
 }
 
 int load_module(void)
 {
-        return ast_custom_function_register(&base64_encode_function) ||
+	return ast_custom_function_register(&base64_encode_function) |
 		ast_custom_function_register(&base64_decode_function);
 }
 
@@ -107,11 +106,3 @@
 {
 	return ASTERISK_GPL_KEY;
 }
-
-/*
-Local Variables:
-mode: C
-c-file-style: "linux"
-indent-tabs-mode: nil
-End:
-*/

Modified: team/rizzo/base/funcs/func_callerid.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/funcs/func_callerid.c?rev=9688&r1=9687&r2=9688&view=diff
==============================================================================
--- team/rizzo/base/funcs/func_callerid.c (original)
+++ team/rizzo/base/funcs/func_callerid.c Sun Feb 12 09:02:06 2006
@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Copyright (C) 1999-2006, Digium, Inc.
  *

[... 3840 lines stripped ...]

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