[asterisk-commits] branch oej/peermatch r14418 - in /team/oej/peermatch: ./ agi/ apps/ cdr/ chan...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Thu Mar 23 11:38:57 MST 2006


Author: oej
Date: Thu Mar 23 12:38:32 2006
New Revision: 14418

URL: http://svn.digium.com/view/asterisk?rev=14418&view=rev
Log:
Reset automerge

Modified:
    team/oej/peermatch/   (props changed)
    team/oej/peermatch/Makefile
    team/oej/peermatch/UPGRADE.txt
    team/oej/peermatch/agi/Makefile
    team/oej/peermatch/apps/app_dial.c
    team/oej/peermatch/apps/app_meetme.c
    team/oej/peermatch/apps/app_voicemail.c
    team/oej/peermatch/astmm.c
    team/oej/peermatch/cdr.c
    team/oej/peermatch/cdr/Makefile
    team/oej/peermatch/channel.c
    team/oej/peermatch/channels/chan_iax2.c
    team/oej/peermatch/channels/chan_misdn.c
    team/oej/peermatch/channels/chan_sip.c
    team/oej/peermatch/channels/chan_skinny.c
    team/oej/peermatch/channels/chan_zap.c
    team/oej/peermatch/channels/misdn/chan_misdn_config.h
    team/oej/peermatch/channels/misdn/isdn_lib.c
    team/oej/peermatch/channels/misdn_config.c
    team/oej/peermatch/codecs/Makefile
    team/oej/peermatch/codecs/gsm/libgsm.vcproj   (props changed)
    team/oej/peermatch/codecs/ilbc/libilbc.vcproj   (props changed)
    team/oej/peermatch/codecs/lpc10/liblpc10.vcproj   (props changed)
    team/oej/peermatch/configs/features.conf.sample
    team/oej/peermatch/configs/indications.conf.sample
    team/oej/peermatch/configs/misdn.conf.sample
    team/oej/peermatch/configs/sip.conf.sample
    team/oej/peermatch/configs/zapata.conf.sample
    team/oej/peermatch/db.c
    team/oej/peermatch/doc/enum.txt
    team/oej/peermatch/editline/term.c
    team/oej/peermatch/funcs/func_enum.c
    team/oej/peermatch/image.c
    team/oej/peermatch/include/asterisk/cdr.h
    team/oej/peermatch/include/asterisk/channel.h
    team/oej/peermatch/include/asterisk/compat.h
    team/oej/peermatch/include/asterisk/module.h
    team/oej/peermatch/include/asterisk/say.h
    team/oej/peermatch/include/solaris-compat/compat.h
    team/oej/peermatch/pbx.c
    team/oej/peermatch/res/res_features.c
    team/oej/peermatch/sample.call
    team/oej/peermatch/say.c
    team/oej/peermatch/sounds.txt
    team/oej/peermatch/strcompat.c

Propchange: team/oej/peermatch/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.

Propchange: team/oej/peermatch/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Propchange: team/oej/peermatch/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu Mar 23 12:38:32 2006
@@ -1,1 +1,1 @@
-/trunk:1-13527
+/trunk:1-14417

Modified: team/oej/peermatch/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/Makefile?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/Makefile (original)
+++ team/oej/peermatch/Makefile Thu Mar 23 12:38:32 2006
@@ -357,7 +357,7 @@
 SUBDIRS=res channels pbx apps codecs formats agi cdr funcs utils stdtime
 
 OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o \
-	translate.o file.o say.o pbx.o cli.o md5.o term.o \
+	translate.o file.o pbx.o cli.o md5.o term.o \
 	ulaw.o alaw.o callerid.o fskmodem.o image.o app.o \
 	cdr.o tdd.o acl.o rtp.o udptl.o manager.o asterisk.o \
 	dsp.o chanvars.o indications.o autoservice.o db.o privacy.o \
@@ -365,6 +365,15 @@
 	utils.o plc.o jitterbuf.o dnsmgr.o devicestate.o \
 	netsock.o slinfactory.o ast_expr2.o ast_expr2f.o \
 	cryptostub.o sha1.o
+
+# we need to link in the objects statically, not as a library, because
+# otherwise modules will not have them available if none of the static
+# objects use it.
+OBJS+= stdtime/localtime.o
+
+# At the moment say.o is an optional component which can be overridden
+# by a module.
+OBJS+= say.o
 
 ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/sys/poll.h),)
   OBJS+= poll.o

Modified: team/oej/peermatch/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/UPGRADE.txt?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/UPGRADE.txt (original)
+++ team/oej/peermatch/UPGRADE.txt Thu Mar 23 12:38:32 2006
@@ -76,3 +76,10 @@
 	ASTBINDIR	/usr/local/bin/asterisk
 	ASTSBINDIR	/usr/local/sbin/asterisk
 
+Sounds:
+
+* The phonetic sounds directory has been removed from the asterisk-sounds package
+  because they are now included directly in Asterisk.  However, it is important to
+  note that the phonetic sounds that existed in asterisk-sounds used a different
+  naming convention than the sounds in Asterisk.  For example, instead of alpha.gsm
+  and bravo.gsm, Asterisk has a_p.gsm and b_p.gsm.

Modified: team/oej/peermatch/agi/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/agi/Makefile?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/agi/Makefile (original)
+++ team/oej/peermatch/agi/Makefile Thu Mar 23 12:38:32 2006
@@ -13,15 +13,11 @@
 
 AGIS=agi-test.agi eagi-test eagi-sphinx-test jukebox.agi
 
-CFLAGS+=
+CFLAGS+=-DNO_AST_MM
 
 LIBS=
 ifeq ($(OSARCH),SunOS)
   LIBS=-lsocket -lnsl ../strcompat.o
-endif
-
-ifeq ($(findstring BSD,${OSARCH}),BSD)
-  CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
 endif
 
 all: depend $(AGIS)

Modified: team/oej/peermatch/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/apps/app_dial.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/apps/app_dial.c (original)
+++ team/oej/peermatch/apps/app_dial.c Thu Mar 23 12:38:32 2006
@@ -112,9 +112,10 @@
 "           other than the number assigned to the caller.\n"
 "    g    - Proceed with dialplan execution at the current extension if the\n"
 "           destination channel hangs up.\n"
-"    G(context^exten^pri) - If the call is answered, transfer both parties to\n"
-"           the specified priority. Optionally, an extension, or extension and\n"
-"           context may be specified. Otherwise, the current extension is used.\n"
+"    G(context^exten^pri) - If the call is answered, transfer the calling party to\n"
+"           the specified priority and the called party to the specified priority+1.\n"
+"           Optionally, an extension, or extension and context may be specified. \n"
+"           Otherwise, the current extension is used.\n"
 "    h    - Allow the called party to hang up by sending the '*' DTMF digit.\n"
 "    H    - Allow the calling party to hang up by hitting the '*' DTMF digit.\n"
 "    j    - Jump to priority n+101 if all of the requested channels were busy.\n"
@@ -1425,6 +1426,7 @@
 			}
 			ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
 			ast_parseable_goto(peer, opt_args[OPT_ARG_GOTO]);
+			peer->priority++;
 			ast_pbx_start(peer);
 			hanguptree(outgoing, NULL);
 			LOCAL_USER_REMOVE(u);

Modified: team/oej/peermatch/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/apps/app_meetme.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/apps/app_meetme.c (original)
+++ team/oej/peermatch/apps/app_meetme.c Thu Mar 23 12:38:32 2006
@@ -957,7 +957,9 @@
 		snprintf(user->namerecloc, sizeof(user->namerecloc),
 			 "%s/meetme/meetme-username-%s-%d", ast_config_AST_SPOOL_DIR,
 			 conf->confno, user->user_no);
-		ast_record_review(chan, "vm-rec-name", user->namerecloc, 10, "sln", &duration, NULL);
+		res = ast_record_review(chan, "vm-rec-name", user->namerecloc, 10, "sln", &duration, NULL);
+		if (res == -1)
+			goto outrun;
 	}
 
 	if ( !(confflags & (CONFFLAG_QUIET | CONFFLAG_NOONLYPERSON)) ) {
@@ -1830,28 +1832,20 @@
 				ast_log(LOG_WARNING, "No %s file :(\n", CONFIG_FILE_NAME);
 				return NULL;
 			}
-			var = ast_variable_browse(cfg, "rooms");
-			for (; var; var = var->next) {
+			for (var = ast_variable_browse(cfg, "rooms"); var; var = var->next) {
 				if (strcasecmp(var->name, "conf"))
 					continue;
 				
 				if (!(parse = ast_strdupa(var->value)))
 					return NULL;
 				
-				AST_STANDARD_APP_ARGS(args, parse);
+				AST_NONSTANDARD_APP_ARGS(args, parse, ',');
 				if (!strcasecmp(args.confno, confno)) {
 					/* Bingo it's a valid conference */
-					if (args.pin) {
-						if (args.pinadmin)
-							cnf = build_conf(args.confno, args.pin, args.pinadmin, make, dynamic, refcount);
-						else
-							cnf = build_conf(args.confno, args.pin, "", make, dynamic, refcount);
-					} else {
-						if (args.pinadmin)
-							cnf = build_conf(args.confno, "", args.pinadmin, make, dynamic, refcount);
-						else
-							cnf = build_conf(args.confno, "", "", make, dynamic, refcount);
-					}
+					cnf = build_conf(args.confno,
+							ast_strlen_zero(args.pin) ? "" : args.pin,
+							ast_strlen_zero(args.pinadmin) ? "" : args.pinadmin,
+							make, dynamic, refcount);
 					break;
 				}
 			}
@@ -2026,7 +2020,7 @@
 									AST_LIST_UNLOCK(&confs);
 									if (!found) {
 										/* At this point, we have a confno_tmp (static conference) that is empty */
-										if ((empty_no_pin && ((!stringp) || (stringp && (stringp[0] == '\0')))) || (!empty_no_pin)) {
+										if ((empty_no_pin && ast_strlen_zero(stringp)) || (!empty_no_pin)) {
 											/* Case 1:  empty_no_pin and pin is nonexistent (NULL)
 											 * Case 2:  empty_no_pin and pin is blank (but not NULL)
 											 * Case 3:  not empty_no_pin

Modified: team/oej/peermatch/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/apps/app_voicemail.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/apps/app_voicemail.c (original)
+++ team/oej/peermatch/apps/app_voicemail.c Thu Mar 23 12:38:32 2006
@@ -2364,7 +2364,7 @@
 
 static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_options *options)
 {
-	char txtfile[256];
+	char tmptxtfile[256], txtfile[256];
 	char callerid[256];
 	FILE *txt;
 	int res = 0;
@@ -2561,7 +2561,8 @@
 				
 			/* Store information */
 			snprintf(txtfile, sizeof(txtfile), "%s.txt", fn);
-			txt = fopen(txtfile, "w+");
+			snprintf(tmptxtfile, sizeof(tmptxtfile), "%s.txt.tmp", fn);
+			txt = fopen(tmptxtfile, "w+");
 			if (txt) {
 				get_date(date, sizeof(date));
 				fprintf(txt, 
@@ -2601,6 +2602,7 @@
 			if (txt) {
 				fprintf(txt, "duration=%d\n", duration);
 				fclose(txt);
+				rename(tmptxtfile, txtfile);
 			}
 				
 			if (duration < vmminmessage) {
@@ -3695,13 +3697,15 @@
 				if (!ast_strlen_zero(prefile)) {
 				/* See if we can find a recorded name for this person instead of their extension number */
 					if (ast_fileexists(prefile, NULL, NULL) > 0) {
-						ast_verbose(VERBOSE_PREFIX_3 "Playing envelope info: CID number '%s' matches mailbox number, playing recorded name\n", callerid);
+						if (option_verbose > 2)
+							ast_verbose(VERBOSE_PREFIX_3 "Playing envelope info: CID number '%s' matches mailbox number, playing recorded name\n", callerid);
 						if (!callback)
 							res = wait_file2(chan, vms, "vm-from");
 						res = ast_streamfile(chan, prefile, chan->language) > -1;
 						res = ast_waitstream(chan, "");
 					} else {
-						ast_verbose(VERBOSE_PREFIX_3 "Playing envelope info: message from '%s'\n", callerid);
+						if (option_verbose > 2)
+							ast_verbose(VERBOSE_PREFIX_3 "Playing envelope info: message from '%s'\n", callerid);
 						/* BB: Say "from extension" as one saying to sound smoother */
 						if (!callback)
 							res = wait_file2(chan, vms, "vm-from-extension");
@@ -5354,7 +5358,8 @@
 					cmd = 't';
 					break;
 				case '2': /* Callback */
-					ast_verbose( VERBOSE_PREFIX_3 "Callback Requested\n");
+					if (option_verbose > 2)
+						ast_verbose( VERBOSE_PREFIX_3 "Callback Requested\n");
 					if (!ast_strlen_zero(vmu->callback) && vms.lastmsg > -1) {
 						cmd = advanced_options(chan, vmu, &vms, vms.curmsg, 2, record_gain);
 						if (cmd == 9) {
@@ -6441,7 +6446,8 @@
 	int retries = 0;
 
 	if (!num) {
-		ast_verbose( VERBOSE_PREFIX_3 "Destination number will be entered manually\n");
+		if (option_verbose > 2)
+			ast_verbose( VERBOSE_PREFIX_3 "Destination number will be entered manually\n");
 		while (retries < 3 && cmd != 't') {
 			destination[1] = '\0';
 			destination[0] = cmd = ast_play_and_wait(chan,"vm-enter-num-to-call");
@@ -6461,7 +6467,8 @@
 				if (cmd < 0)
 					return 0;
 				if (cmd == '*') {
-					ast_verbose( VERBOSE_PREFIX_3 "User hit '*' to cancel outgoing call\n");
+					if (option_verbose > 2)
+						ast_verbose( VERBOSE_PREFIX_3 "User hit '*' to cancel outgoing call\n");
 					return 0;
 				}
 				if ((cmd = ast_readstring(chan,destination + strlen(destination),sizeof(destination)-1,6000,10000,"#")) < 0) 
@@ -6475,14 +6482,16 @@
 		}
 		
 	} else {
-		ast_verbose( VERBOSE_PREFIX_3 "Destination number is CID number '%s'\n", num);
+		if (option_verbose > 2)
+			ast_verbose( VERBOSE_PREFIX_3 "Destination number is CID number '%s'\n", num);
 		ast_copy_string(destination, num, sizeof(destination));
 	}
 
 	if (!ast_strlen_zero(destination)) {
 		if (destination[strlen(destination) -1 ] == '*')
 			return 0; 
-		ast_verbose( VERBOSE_PREFIX_3 "Placing outgoing call to extension '%s' in context '%s' from context '%s'\n", destination, outgoing_context, chan->context);
+		if (option_verbose > 2)
+			ast_verbose( VERBOSE_PREFIX_3 "Placing outgoing call to extension '%s' in context '%s' from context '%s'\n", destination, outgoing_context, chan->context);
 		ast_copy_string(chan->exten, destination, sizeof(chan->exten));
 		ast_copy_string(chan->context, outgoing_context, sizeof(chan->context));
 		chan->priority = 0;
@@ -6553,7 +6562,8 @@
 							if (res)
 								return 9;
 						} else {
-							ast_verbose( VERBOSE_PREFIX_3 "Caller can not specify callback number - no dialout context available\n");
+							if (option_verbose > 2)
+								ast_verbose( VERBOSE_PREFIX_3 "Caller can not specify callback number - no dialout context available\n");
 							res = ast_play_and_wait(chan, "vm-sorry");
 						}
 						return res;
@@ -6574,7 +6584,8 @@
 						break;
 					default:
 						if (num) {
-							ast_verbose( VERBOSE_PREFIX_3 "Confirm CID number '%s' is number to use for callback\n", num);
+							if (option_verbose > 2)
+								ast_verbose( VERBOSE_PREFIX_3 "Confirm CID number '%s' is number to use for callback\n", num);
 							res = ast_play_and_wait(chan, "vm-num-i-have");
 							if (!res)
 								res = play_message_callerid(chan, vms, num, vmu->context, 1);
@@ -6616,7 +6627,8 @@
 		if (!ast_strlen_zero(cid)) {
 			ast_callerid_parse(cid, &name, &num);
 			if (!num) {
-				ast_verbose(VERBOSE_PREFIX_3 "No CID number available, no reply sent\n");
+				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "No CID number available, no reply sent\n");
 				if (!res)
 					res = ast_play_and_wait(chan, "vm-nonumber");
 				return res;
@@ -6624,7 +6636,8 @@
 				if (find_user(NULL, vmu->context, num)) {
 					struct leave_vm_options leave_options;
 
-					ast_verbose(VERBOSE_PREFIX_3 "Leaving voicemail for '%s' in context '%s'\n", num, vmu->context);
+					if (option_verbose > 2)
+						ast_verbose(VERBOSE_PREFIX_3 "Leaving voicemail for '%s' in context '%s'\n", num, vmu->context);
 					
 					memset(&leave_options, 0, sizeof(leave_options));
 					leave_options.record_gain = record_gain;
@@ -6634,7 +6647,8 @@
 					return res;
 				} else {
 					/* Sender has no mailbox, can't reply */
-					ast_verbose( VERBOSE_PREFIX_3 "No mailbox number '%s' in context '%s', no reply sent\n", num, vmu->context);
+					if (option_verbose > 2)
+						ast_verbose( VERBOSE_PREFIX_3 "No mailbox number '%s' in context '%s', no reply sent\n", num, vmu->context);
 					ast_play_and_wait(chan, "vm-nobox");
 					res = 't';
 					return res;
@@ -6685,7 +6699,8 @@
  				break;
  			} else {
  				/* Otherwise 1 is to save the existing message */
- 				ast_verbose(VERBOSE_PREFIX_3 "Saving message as is\n");
+ 				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "Saving message as is\n");
  				ast_streamfile(chan, "vm-msgsaved", chan->language);
  				ast_waitstream(chan, "");
 				STORE(recordfile, vmu->mailbox, vmu->context, -1);
@@ -6695,17 +6710,21 @@
  			}
  		case '2':
  			/* Review */
- 			ast_verbose(VERBOSE_PREFIX_3 "Reviewing the message\n");
+ 			if (option_verbose > 2)
+				ast_verbose(VERBOSE_PREFIX_3 "Reviewing the message\n");
  			ast_streamfile(chan, recordfile, chan->language);
  			cmd = ast_waitstream(chan, AST_DIGIT_ANY);
  			break;
  		case '3':
  			message_exists = 0;
  			/* Record */
- 			if (recorded == 1)
-				ast_verbose(VERBOSE_PREFIX_3 "Re-recording the message\n");
- 			else	
-				ast_verbose(VERBOSE_PREFIX_3 "Recording the message\n");
+ 			if (recorded == 1) {
+				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "Re-recording the message\n");
+ 			} else {	
+				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "Recording the message\n");
+			}
 			if (recorded && outsidecaller) {
  				cmd = ast_play_and_wait(chan, INTRO);
  				cmd = ast_play_and_wait(chan, "beep");
@@ -6729,14 +6748,16 @@
 #if 0			
  			else if (vmu->review && (*duration < 5)) {
  				/* Message is too short */
- 				ast_verbose(VERBOSE_PREFIX_3 "Message too short\n");
+ 				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "Message too short\n");
 				cmd = ast_play_and_wait(chan, "vm-tooshort");
  				cmd = vm_delete(recordfile);
  				break;
  			}
  			else if (vmu->review && (cmd == 2 && *duration < (maxsilence + 3))) {
  				/* Message is all silence */
- 				ast_verbose(VERBOSE_PREFIX_3 "Nothing recorded\n");
+ 				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "Nothing recorded\n");
  				cmd = vm_delete(recordfile);
 				cmd = ast_play_and_wait(chan, "vm-nothingrecorded");
 				if (!cmd)

Modified: team/oej/peermatch/astmm.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/astmm.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/astmm.c (original)
+++ team/oej/peermatch/astmm.c Thu Mar 23 12:38:32 2006
@@ -38,6 +38,7 @@
 #include "asterisk/options.h"
 #include "asterisk/lock.h"
 #include "asterisk/strings.h"
+#include "asterisk/unaligned.h"
 
 #define SOME_PRIME 563
 
@@ -104,7 +105,7 @@
 		regions[hash] = reg;
 		reg->fence = FENCE_MAGIC;
 		fence = (ptr + reg->len);
-		*fence = FENCE_MAGIC;
+		put_unaligned_uint32(fence, FENCE_MAGIC);
 	}
 	ast_mutex_unlock(&reglock);
 	if (!reg) {
@@ -166,7 +167,7 @@
 				fflush(mmlog);
 			}
 		}
-		if (*fence != FENCE_MAGIC) {
+		if (get_unaligned_uint32(fence) != FENCE_MAGIC) {
 			fprintf(stderr, "WARNING: High fence violation at %p, in %s of %s, line %d\n", reg->data, reg->func, reg->file, reg->lineno);
 			if (mmlog) {
 				fprintf(mmlog, "%ld - WARNING: High fence violation at %p, in %s of %s, line %d\n", time(NULL), reg->data, reg->func, reg->file, reg->lineno);
@@ -316,7 +317,7 @@
 						fflush(mmlog);
 					}
 				}
-				if (*fence != FENCE_MAGIC) {
+				if (get_unaligned_uint32(fence) != FENCE_MAGIC) {
 					fprintf(stderr, "WARNING: High fence violation at %p, in %s of %s, line %d\n", reg->data, reg->func, reg->file, reg->lineno);
 					if (mmlog) {
 						fprintf(mmlog, "%ld - WARNING: High fence violation at %p, in %s of %s, line %d\n", time(NULL), reg->data, reg->func, reg->file, reg->lineno);

Modified: team/oej/peermatch/cdr.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/cdr.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/cdr.c (original)
+++ team/oej/peermatch/cdr.c Thu Mar 23 12:38:32 2006
@@ -515,8 +515,10 @@
 		chan = !ast_strlen_zero(cdr->channel) ? cdr->channel : "<unknown>";
 		if (ast_test_flag(cdr, AST_CDR_FLAG_POSTED))
 			ast_log(LOG_WARNING, "CDR on channel '%s' already posted\n", chan);
-		if (!ast_test_flag(cdr, AST_CDR_FLAG_LOCKED))
-			cdr->disposition = AST_CDR_FAILED;
+		if (!ast_test_flag(cdr, AST_CDR_FLAG_LOCKED)) {
+			if (cdr->disposition < AST_CDR_FAILED)
+				cdr->disposition = AST_CDR_FAILED;
+		}
 		cdr = cdr->next;
 	}
 }

Modified: team/oej/peermatch/cdr/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/cdr/Makefile?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/cdr/Makefile (original)
+++ team/oej/peermatch/cdr/Makefile Thu Mar 23 12:38:32 2006
@@ -35,11 +35,6 @@
   CYGSOLIB=-L.. -L. -lasterisk.dll
 else
   CFLAGS+=-fPIC
-endif
-
-ifeq ($(findstring BSD,${OSARCH}),BSD)
-  CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include
-  SOLINK+=-L$(CROSS_COMPILE_TARGET)/usr/local/lib
 endif
 
 #The problem with sparc is the best stuff is in newer versions of gcc (post 3.0) only.

Modified: team/oej/peermatch/channel.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/channel.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/channel.c (original)
+++ team/oej/peermatch/channel.c Thu Mar 23 12:38:32 2006
@@ -1006,6 +1006,7 @@
 	while ((vardata = AST_LIST_REMOVE_HEAD(headp, entries)))
 		ast_var_delete(vardata);
 
+	ast_string_field_free_all(chan);
 	free(chan);
 	AST_LIST_UNLOCK(&channels);
 

Modified: team/oej/peermatch/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/channels/chan_iax2.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/channels/chan_iax2.c (original)
+++ team/oej/peermatch/channels/chan_iax2.c Thu Mar 23 12:38:32 2006
@@ -5037,7 +5037,8 @@
 				ast_copy_string(iaxs[callno]->cid_name, user->cid_name, sizeof(iaxs[callno]->cid_name));
 				iaxs[callno]->calling_pres = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
 			}
-			ast_copy_string(iaxs[callno]->ani, user->cid_num, sizeof(iaxs[callno]->ani));
+			if (ast_strlen_zero(iaxs[callno]->ani))
+				ast_copy_string(iaxs[callno]->ani, user->cid_num, sizeof(iaxs[callno]->ani));
 		} else {
 			iaxs[callno]->calling_pres = AST_PRES_NUMBER_NOT_AVAILABLE;
 		}

Modified: team/oej/peermatch/channels/chan_misdn.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/channels/chan_misdn.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/channels/chan_misdn.c (original)
+++ team/oej/peermatch/channels/chan_misdn.c Thu Mar 23 12:38:32 2006
@@ -1105,6 +1105,21 @@
 	
 	chan_misdn_log(1,port,"update_config: Getting Config\n");
 
+
+	int hdlc=0;
+	misdn_cfg_get( port, MISDN_CFG_HDLC, &hdlc, sizeof(int));
+	
+	if (hdlc) {
+		switch (bc->capability) {
+		case INFO_CAPABILITY_DIGITAL_UNRESTRICTED:
+		case INFO_CAPABILITY_DIGITAL_RESTRICTED:
+			chan_misdn_log(1,bc->port," --> CONF HDLC\n");
+			bc->hdlc=1;
+			break;
+		}
+		
+	}
+	
 	
 	int pres, screen;
 			
@@ -1244,7 +1259,19 @@
 
 	misdn_cfg_get( port, MISDN_CFG_NEED_MORE_INFOS, &bc->need_more_infos, sizeof(int));
 	
-
+	int hdlc=0;
+	misdn_cfg_get( port, MISDN_CFG_HDLC, &hdlc, sizeof(int));
+	
+	if (hdlc) {
+		switch (bc->capability) {
+		case INFO_CAPABILITY_DIGITAL_UNRESTRICTED:
+		case INFO_CAPABILITY_DIGITAL_RESTRICTED:
+			chan_misdn_log(1,bc->port," --> CONF HDLC\n");
+			bc->hdlc=1;
+			break;
+		}
+		
+	}
 	/*Initialize new Jitterbuffer*/
 	{
 		misdn_cfg_get( port, MISDN_CFG_JITTERBUFFER, &ch->jb_len, sizeof(int));
@@ -1573,7 +1600,7 @@
 
 		/* update screening and presentation */ 
 		update_config(ch,ORG_AST);
-
+		
 		/* fill in some ies from channel vary*/
 		import_ies(ast, newbc);
 		
@@ -2053,7 +2080,7 @@
 
 
 	if ( !frame->subclass) {
-		chan_misdn_log(0, ch->bc->port, "misdn_write: * prods us\n");
+		chan_misdn_log(2, ch->bc->port, "misdn_write: * prods us\n");
 		return 0;
 	}
 	
@@ -2137,19 +2164,19 @@
 		int ecwb;
 		misdn_cfg_get( ch1->bc->port, MISDN_CFG_ECHOCANCELWHENBRIDGED, &ecwb, sizeof(int));
 		if ( !ecwb ) {
-			chan_misdn_log(0, ch1->bc->port, "Disabling Echo Cancellor when Bridged\n");
+			chan_misdn_log(2, ch1->bc->port, "Disabling Echo Cancellor when Bridged\n");
 			ch1->bc->ec_enable=0;
 			manager_ec_disable(ch1->bc);
 		}
 		misdn_cfg_get( ch2->bc->port, MISDN_CFG_ECHOCANCELWHENBRIDGED, &ecwb, sizeof(int));
 		if ( !ecwb ) {
-			chan_misdn_log(0, ch2->bc->port, "Disabling Echo Cancellor when Bridged\n");
+			chan_misdn_log(2, ch2->bc->port, "Disabling Echo Cancellor when Bridged\n");
 			ch2->bc->ec_enable=0;
 			manager_ec_disable(ch2->bc);
 		}
 		
 		/* trying to make a mISDN_dsp conference */
-		chan_misdn_log(0, ch1->bc->port, "I SEND: Making conference with Number:%d\n", (ch1->bc->pid<<1) +1);
+		chan_misdn_log(1, ch1->bc->port, "I SEND: Making conference with Number:%d\n", (ch1->bc->pid<<1) +1);
 
 		misdn_lib_bridge(ch1->bc,ch2->bc);
 	}
@@ -2201,7 +2228,7 @@
 	const struct tone_zone_sound *ts= NULL;
 	struct ast_channel *ast=cl->ast;
 	
-	chan_misdn_log(0,cl->bc->port,"Tone Indicate:\n");
+	chan_misdn_log(2,cl->bc->port,"Tone Indicate:\n");
 	
 	if (!cl->ast) {
 		return 0;
@@ -2209,17 +2236,17 @@
 	
 	switch (tone) {
 	case TONE_DIAL:
-		chan_misdn_log(0,cl->bc->port," --> Dial\n");
+		chan_misdn_log(2,cl->bc->port," --> Dial\n");
 		ts=ast_get_indication_tone(ast->zone,"dial");
 		misdn_lib_tone_generator_start(cl->bc);
 		break;
 	case TONE_ALERTING:
-		chan_misdn_log(0,cl->bc->port," --> Ring\n");
+		chan_misdn_log(2,cl->bc->port," --> Ring\n");
 		ts=ast_get_indication_tone(ast->zone,"ring");
 		misdn_lib_tone_generator_stop(cl->bc);
 		break;
 	case TONE_BUSY:
-		chan_misdn_log(0,cl->bc->port," --> Busy\n");
+		chan_misdn_log(2,cl->bc->port," --> Busy\n");
 		ts=ast_get_indication_tone(ast->zone,"busy");
 		misdn_lib_tone_generator_stop(cl->bc);
 		break;
@@ -2227,7 +2254,7 @@
 		break;
 
 	case TONE_NONE:
-		chan_misdn_log(0,cl->bc->port," --> None\n");
+		chan_misdn_log(2,cl->bc->port," --> None\n");
 		misdn_lib_tone_generator_stop(cl->bc);
 		ast_playtones_stop(ast);
 		break;
@@ -3298,7 +3325,7 @@
 		}
 
 		
-			chan_misdn_log(0,bc->port,"CONTEXT:%s\n",ch->context);
+			chan_misdn_log(5,bc->port,"CONTEXT:%s\n",ch->context);
 			if(!ast_canmatch_extension(ch->ast, ch->context, bc->dad, 1, bc->oad)) {
 			
 			chan_misdn_log(-1, bc->port, "Extension can never match, so disconnecting\n");

Modified: team/oej/peermatch/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/channels/chan_sip.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/channels/chan_sip.c (original)
+++ team/oej/peermatch/channels/chan_sip.c Thu Mar 23 12:38:32 2006
@@ -396,7 +396,7 @@
 static struct ast_codec_pref default_prefs;		/*!< Default codec prefs */
 
 /* Global settings only apply to the channel */
-static int global_rtautoclear = 120;
+static int global_rtautoclear;
 static int global_notifyringing;	/*!< Send notifications on ringing */
 static int srvlookup;			/*!< SRV Lookup on or off. Default is off, RFC behavior is on */
 static int pedanticsipchecking;		/*!< Extra checking ?  Default off */
@@ -415,7 +415,7 @@
 static int recordhistory;		/*!< Record SIP history. Off by default */
 static int dumphistory;			/*!< Dump history to verbose before destroying SIP dialog */
 static char global_realm[MAXHOSTNAMELEN]; 		/*!< Default realm */
-static char regcontext[AST_MAX_CONTEXT];		/*!< Context for auto-extensions */
+static char global_regcontext[AST_MAX_CONTEXT];		/*!< Context for auto-extensions */
 static char global_useragent[AST_MAX_EXTENSION];	/*!< Useragent for the SIP channel */
 static int allow_external_domains;	/*!< Accept calls to external SIP domains? */
 static int global_callevents;		/*!< Whether we send manager events or not */
@@ -599,7 +599,7 @@
 #define SIP_FLAGS_TO_COPY \
 	(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
 	 SIP_PROG_INBAND | SIP_OSPAUTH | SIP_USECLIENTCODE | SIP_NAT | \
-	 SIP_INSECURE_PORT | SIP_INSECURE_INVITE)
+	 SIP_USEREQPHONE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE)
 
 /* a new page of flags for peers */
 #define SIP_PAGE2_RTCACHEFRIENDS	(1 << 0)
@@ -620,7 +620,6 @@
 #define sipdebug		ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG)
 #define sipdebug_config		ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONFIG)
 #define sipdebug_console	ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONSOLE)
-
 
 /*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe  */
 static struct sip_pvt {
@@ -1648,15 +1647,15 @@
 {
 	char multi[256];
 	char *stringp, *ext;
-	if (!ast_strlen_zero(regcontext)) {
+	if (!ast_strlen_zero(global_regcontext)) {
 		ast_copy_string(multi, ast_strlen_zero(peer->regexten) ? peer->name : peer->regexten, sizeof(multi));
 		stringp = multi;
 		while((ext = strsep(&stringp, "&"))) {
 			if (onoff)
-				ast_add_extension(regcontext, 1, ext, 1, NULL, NULL, "Noop",
+				ast_add_extension(global_regcontext, 1, ext, 1, NULL, NULL, "Noop",
 						  ast_strdup(peer->name), free, "SIP");
 			else
-				ast_context_remove_extension(regcontext, ext, 1, NULL);
+				ast_context_remove_extension(global_regcontext, ext, 1, NULL);
 		}
 	}
 }
@@ -1683,6 +1682,7 @@
 	if (device->type & SIP_PEER) {
 		register_peer_exten(device, FALSE);
 		clear_realm_authentication(device->auth);
+		device->auth = (struct sip_auth *) NULL;
 		if (ast_test_flag((&device->flags_page2), SIP_PAGE2_SELFDESTRUCT))
 			apeerobjs--;
 		else if (ast_test_flag(device, SIP_REALTIME))
@@ -1696,7 +1696,6 @@
 		else
 			suserobjs--;
 	}
-	device->auth = (struct sip_auth *) NULL;
 	if (device->dnsmgr)
 		ast_dnsmgr_release(device->dnsmgr);
 	free(device);
@@ -5750,7 +5749,7 @@
 	destroy_association(peer);
 	
 	manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
-	register_peer_exten(peer, 0);
+	register_peer_exten(peer, FALSE);
 	peer->expire = -1;
 	ast_device_state_changed("SIP/%s", peer->name);
 	if (ast_test_flag((&peer->flags_page2), SIP_PAGE2_SELFDESTRUCT) || ast_test_flag((&peer->flags_page2), SIP_PAGE2_RTAUTOCLEAR)) {
@@ -5829,7 +5828,7 @@
 	if (peer->expire > -1)
 		ast_sched_del(sched, peer->expire);
 	peer->expire = ast_sched_add(sched, (expiry + 10) * 1000, expire_register, peer);
-	register_peer_exten(peer, 1);
+	register_peer_exten(peer, TRUE);
 }
 
 /*! \brief Parse contact header for 200 OK on INVITE */
@@ -7904,7 +7903,6 @@
 
 static char mandescr_show_peer[] = 
 "Description: Show one SIP peer with details on current status.\n"
-"  The XML format is under development, feedback welcome! /oej\n"
 "Variables: \n"
 "  Peer: <name>           The peer name you want to check.\n"
 "  ActionID: <id>	  Optional action ID for this AMI transaction.\n";
@@ -8018,6 +8016,8 @@
 		ast_cli(fd, "  ToHost       : %s\n", peer->tohost);
 		ast_cli(fd, "  Addr->IP     : %s Port %d\n",  peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port));
 		ast_cli(fd, "  Defaddr->IP  : %s Port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
+		if (!ast_strlen_zero(global_regcontext))
+			ast_cli(fd, "  Reg. exten   : %s\n", peer->regexten);
 		ast_cli(fd, "  Def. Username: %s\n", peer->username);
 		ast_cli(fd, "  SIP Options  : ");
 		if (peer->sipoptions) {
@@ -8093,6 +8093,8 @@
 		ast_cli(fd, "Address-IP: %s\r\nAddress-Port: %d\r\n",  peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "", ntohs(peer->addr.sin_port));
 		ast_cli(fd, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
 		ast_cli(fd, "Default-Username: %s\r\n", peer->username);
+		if (!ast_strlen_zero(global_regcontext))
+			ast_cli(fd, "RegExtension: %s\r\n", peer->regexten);
 		ast_cli(fd, "Codecs: ");
 		ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
 		ast_cli(fd, "%s\r\n", codec_buf);
@@ -8242,7 +8244,7 @@
 	ast_cli(fd, "  Realm. auth:            %s\n", authl ? "Yes": "No");
 	ast_cli(fd, "  User Agent:             %s\n", global_useragent);
 	ast_cli(fd, "  MWI checking interval:  %d secs\n", global_mwitime);
-	ast_cli(fd, "  Reg. context:           %s\n", ast_strlen_zero(regcontext) ? "(not set)" : regcontext);
+	ast_cli(fd, "  Reg. context:           %s\n", ast_strlen_zero(global_regcontext) ? "(not set)" : global_regcontext);
 	ast_cli(fd, "  Caller ID:              %s\n", default_callerid);
 	ast_cli(fd, "  From: Domain:           %s\n", default_fromdomain);
 	ast_cli(fd, "  Record SIP history:     %s\n", recordhistory ? "On" : "Off");
@@ -10320,6 +10322,7 @@
 		if (required_profile) { 	/* They require something */
 			/* At this point we support no extensions, so fail */
 			transmit_response_with_unsupported(p, "420 Bad extension", req, required);
+			ast_log(LOG_WARNING,"Received SIP INVITE with unsupported required extension: %s\n", required);
 			if (!p->lastinvite)
 				ast_set_flag(p, SIP_NEEDDESTROY);	
 			return -1;
@@ -12333,7 +12336,7 @@
 
 	/* Reset channel settings to default before re-configuring */
 	allow_external_domains = DEFAULT_ALLOW_EXT_DOM;				/* Allow external invites */
-	regcontext[0] = '\0';
+	global_regcontext[0] = '\0';
 	expiry = DEFAULT_EXPIRY;
 	global_notifyringing = DEFAULT_NOTIFYRINGING;
 	ast_copy_string(global_useragent, DEFAULT_USERAGENT, sizeof(global_useragent));
@@ -12351,6 +12354,7 @@
 	global_rtptimeout = 0;
 	global_rtpholdtimeout = 0;
 	global_rtpkeepalive = 0;
+	global_rtautoclear = 120;
 	ast_set_flag(&global_flags_page2, SIP_PAGE2_RTUPDATE);
 
 	/* Initialize some reasonable defaults at SIP reload (used both for channel and as default for peers and users */
@@ -12442,10 +12446,10 @@
 		} else if (!strcasecmp(v->name, "language")) {
 			ast_copy_string(default_language, v->value, sizeof(default_language));
 		} else if (!strcasecmp(v->name, "regcontext")) {
-			ast_copy_string(regcontext, v->value, sizeof(regcontext));
+			ast_copy_string(global_regcontext, v->value, sizeof(global_regcontext));
 			/* Create context if it doesn't exist already */
-			if (!ast_context_find(regcontext))
-				ast_context_create(NULL, regcontext, "SIP");
+			if (!ast_context_find(global_regcontext))
+				ast_context_create(NULL, global_regcontext, "SIP");
 		} else if (!strcasecmp(v->name, "callerid")) {
 			ast_copy_string(default_callerid, v->value, sizeof(default_callerid));
 		} else if (!strcasecmp(v->name, "fromdomain")) {

Modified: team/oej/peermatch/channels/chan_skinny.c
URL: http://svn.digium.com/view/asterisk/team/oej/peermatch/channels/chan_skinny.c?rev=14418&r1=14417&r2=14418&view=diff
==============================================================================
--- team/oej/peermatch/channels/chan_skinny.c (original)
+++ team/oej/peermatch/channels/chan_skinny.c Thu Mar 23 12:38:32 2006
@@ -19,7 +19,7 @@
 /*! \file
  *
  * \brief Implementation of the Skinny protocol
- * 
+ *
  * \author Jeremy McNamara & Florian Overkamp
  * \ingroup channel_drivers
  */
@@ -70,9 +70,9 @@
 #include "asterisk/dsp.h"
 #include "asterisk/stringfields.h"
 
-/************************************************************************************/
-/*                         Skinny/Asterisk Protocol Settings                        */
-/************************************************************************************/
+/*************************************
+ * Skinny/Asterisk Protocol Settings *
+ *************************************/
 static const char desc[] = "Skinny Client Control Protocol (Skinny)";
 static const char tdesc[] = "Skinny Client Control Protocol (Skinny)";
 static const char config[] = "skinny.conf";
@@ -81,17 +81,17 @@
 static int capability = AST_FORMAT_ULAW;
 
 #define DEFAULT_SKINNY_PORT	2000
-#define DEFAULT_SKINNY_BACKLOG  2
+#define DEFAULT_SKINNY_BACKLOG	2
 #define SKINNY_MAX_PACKET	1000
 
-static int  keep_alive = 120;
+static int keep_alive = 120;
 static char date_format[6] = "D-M-Y";
 static char version_id[16] = "P002F202";
 
 /* these should be in an include file, but dunno what to include */
-typedef unsigned char  UINT8;
-typedef unsigned short UINT16;
-typedef unsigned int   UINT32;
+typedef unsigned char	UINT8;
+typedef unsigned short	UINT16;
+typedef unsigned int	UINT32;
 
 #if __BYTE_ORDER == __LITTLE_ENDIAN
 #define letohl(x) (x)
@@ -101,13 +101,13 @@
 #else
 #if defined(SOLARIS) || defined(__Darwin__) || defined(__NetBSD__)
 #define __bswap_16(x) \
-     ((((x) & 0xff00) >> 8) | \
-      (((x) & 0x00ff) << 8))
+	((((x) & 0xff00) >> 8) | \
+	 (((x) & 0x00ff) << 8))
 #define __bswap_32(x) \
-     ((((x) & 0xff000000) >> 24) | \
-      (((x) & 0x00ff0000) >>  8) | \
-      (((x) & 0x0000ff00) <<  8) | \
-      (((x) & 0x000000ff) << 24))
+	((((x) & 0xff000000) >> 24) | \
+	 (((x) & 0x00ff0000) >>  8) | \
+	 (((x) & 0x0000ff00) <<  8) | \
+	 (((x) & 0x000000ff) << 24))
 #else
 #include <bits/byteswap.h>
 #endif
@@ -118,9 +118,9 @@
 #endif
 
 
-/************************************************************************************/
-/*                                Protocol Messages                                 */
-/************************************************************************************/
+/*********************
+ * Protocol Messages *
+ *********************/
 /* message types */
 #define	KEEP_ALIVE_MESSAGE 0x0000
 /* no additional struct */
@@ -147,18 +147,18 @@
 	int stimulus;
 	int stimulusInstance;
 } stimulus_message;
-		
+
 #define OFFHOOK_MESSAGE 0x0006
 #define ONHOOK_MESSAGE 0x0007
 
 #define	CAPABILITIES_RES_MESSAGE 0x0010
-typedef struct station_capabilities {	
+typedef struct station_capabilities {
 	int codec;
 	int frames;
 	union {
 		char res[8];
 		long rate;
-	} payloads;	
+	} payloads;
 } station_capabilities;
 
 typedef struct capabilities_res_message {
@@ -182,7 +182,7 @@
 #define SERVER_REQUEST_MESSAGE 0x0012
 #define ALARM_MESSAGE 0x0020
 
-#define OPEN_RECIEVE_CHANNEL_ACK_MESSAGE 0x0022	
+#define OPEN_RECIEVE_CHANNEL_ACK_MESSAGE 0x0022
 typedef struct open_recieve_channel_ack_message {
 	int status;
 	char ipAddr[4];
@@ -222,7 +222,7 @@
 	int deviceStimulus;
 } set_lamp_message;
 
-#define SET_SPEAKER_MESSAGE 0x0088 
+#define SET_SPEAKER_MESSAGE 0x0088
 typedef struct set_speaker_message {
 	int mode;
 } set_speaker_message;
@@ -248,7 +248,7 @@
 #define STOP_MEDIA_TRANSMISSION_MESSAGE 0x008B
 typedef struct stop_media_transmission_message {
 	int conferenceId;
-        int passThruPartyId;
+	int passThruPartyId;
 } stop_media_transmission_message;
 
 #define CALL_INFO_MESSAGE 0x008F
@@ -257,9 +257,9 @@
 	char callingParty[24];
 	char calledPartyName[40];
 	char calledParty[24];
-	int  instance;
-	int  reference;
-	int  type;
+	int instance;
+	int reference;
+	int type;
 	char originalCalledPartyName[40];
 	char originalCalledParty[24];
 } call_info_message;
@@ -273,10 +273,10 @@
 
 #define LINE_STAT_RES_MESSAGE 0x0092
 typedef struct line_stat_res_message {
-	int  linenumber;
+	int linenumber;
 	char lineDirNumber[24];
 	char lineDisplayName[42];
-	int  space;
+	int space;
 } line_stat_res_message;
 
 #define DEFINETIMEDATE_MESSAGE 0x0094
@@ -291,7 +291,7 @@
 	int milliseconds;
 	int timestamp;
 } definetimedate_message;
- 
+
 #define DISPLAYTEXT_MESSAGE 0x0099
 typedef struct displaytext_message {

[... 3764 lines stripped ...]


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