[asterisk-commits] branch oej/siptransfer r17987 - in /team/oej/siptransfer: ./ apps/ channels/ ...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Thu Apr 6 12:21:37 MST 2006


Author: oej
Date: Thu Apr  6 14:21:00 2006
New Revision: 17987

URL: http://svn.digium.com/view/asterisk?rev=17987&view=rev
Log:
- Update to trunk
- Expand README with development notes

Added:
    team/oej/siptransfer/res/res_config_pgsql.c
      - copied unchanged from r17740, trunk/res/res_config_pgsql.c
Removed:
    team/oej/siptransfer/formats/format_au.c
    team/oej/siptransfer/formats/format_pcm_alaw.c
Modified:
    team/oej/siptransfer/   (props changed)
    team/oej/siptransfer/.cleancount
    team/oej/siptransfer/CREDITS
    team/oej/siptransfer/Makefile
    team/oej/siptransfer/README.siptransfer
    team/oej/siptransfer/app.c
    team/oej/siptransfer/apps/app_alarmreceiver.c
    team/oej/siptransfer/apps/app_channelredirect.c
    team/oej/siptransfer/apps/app_exec.c
    team/oej/siptransfer/apps/app_page.c
    team/oej/siptransfer/apps/app_queue.c
    team/oej/siptransfer/apps/app_senddtmf.c
    team/oej/siptransfer/apps/app_voicemail.c
    team/oej/siptransfer/channel.c
    team/oej/siptransfer/channels/chan_agent.c
    team/oej/siptransfer/channels/chan_h323.c
    team/oej/siptransfer/channels/chan_iax2.c
    team/oej/siptransfer/channels/chan_local.c
    team/oej/siptransfer/channels/chan_mgcp.c
    team/oej/siptransfer/channels/chan_misdn.c
    team/oej/siptransfer/channels/chan_sip.c
    team/oej/siptransfer/channels/chan_zap.c
    team/oej/siptransfer/channels/h323/ast_h323.cpp
    team/oej/siptransfer/channels/misdn/Makefile
    team/oej/siptransfer/channels/misdn/isdn_lib.c
    team/oej/siptransfer/channels/misdn/isdn_lib.h
    team/oej/siptransfer/channels/misdn/isdn_msg_parser.c
    team/oej/siptransfer/channels/misdn/portinfo.c
    team/oej/siptransfer/configs/features.conf.sample
    team/oej/siptransfer/configs/sip.conf.sample
    team/oej/siptransfer/devicestate.c
    team/oej/siptransfer/dns.c
    team/oej/siptransfer/doc/CODING-GUIDELINES
    team/oej/siptransfer/doc/manager.txt
    team/oej/siptransfer/enum.c
    team/oej/siptransfer/file.c
    team/oej/siptransfer/formats/Makefile
    team/oej/siptransfer/formats/format_g723.c
    team/oej/siptransfer/formats/format_g726.c
    team/oej/siptransfer/formats/format_g729.c
    team/oej/siptransfer/formats/format_gsm.c
    team/oej/siptransfer/formats/format_h263.c
    team/oej/siptransfer/formats/format_h264.c
    team/oej/siptransfer/formats/format_ilbc.c
    team/oej/siptransfer/formats/format_ogg_vorbis.c
    team/oej/siptransfer/formats/format_pcm.c
    team/oej/siptransfer/formats/format_sln.c
    team/oej/siptransfer/formats/format_vox.c
    team/oej/siptransfer/formats/format_wav.c
    team/oej/siptransfer/formats/format_wav_gsm.c
    team/oej/siptransfer/http.c
    team/oej/siptransfer/include/asterisk/channel.h
    team/oej/siptransfer/include/asterisk/doxyref.h
    team/oej/siptransfer/include/asterisk/file.h
    team/oej/siptransfer/include/asterisk/http.h
    team/oej/siptransfer/include/asterisk/linkedlists.h
    team/oej/siptransfer/include/asterisk/pbx.h
    team/oej/siptransfer/manager.c
    team/oej/siptransfer/pbx.c
    team/oej/siptransfer/pbx/pbx_dundi.c
    team/oej/siptransfer/res/Makefile
    team/oej/siptransfer/res/res_musiconhold.c
    team/oej/siptransfer/rtp.c
    team/oej/siptransfer/udptl.c

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

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

Propchange: team/oej/siptransfer/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu Apr  6 14:21:00 2006
@@ -1,1 +1,1 @@
-/trunk:1-17102
+/trunk:1-17742

Modified: team/oej/siptransfer/.cleancount
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/.cleancount?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/.cleancount (original)
+++ team/oej/siptransfer/.cleancount Thu Apr  6 14:21:00 2006
@@ -1,1 +1,1 @@
-12
+13

Modified: team/oej/siptransfer/CREDITS
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/CREDITS?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/CREDITS (original)
+++ team/oej/siptransfer/CREDITS Thu Apr  6 14:21:00 2006
@@ -12,6 +12,8 @@
 Telesthetic - for supporting SIP development
 
 Christos Ricudis - for substantial code contributions
+
+nic.at - ENUM support in Asterisk
 
 Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development
 

Modified: team/oej/siptransfer/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/Makefile?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/Makefile (original)
+++ team/oej/siptransfer/Makefile Thu Apr  6 14:21:00 2006
@@ -28,6 +28,7 @@
 ifeq ($(CROSS_COMPILE),)
   OSARCH=$(shell uname -s)
   OSREV=$(shell uname -r)
+  MARCH=$(shell uname -m)
 else
   OSARCH=$(CROSS_ARCH)
   OSREV=$(CROSS_REV)
@@ -397,8 +398,11 @@
   AUDIO_LIBS=-framework CoreAudio
   ASTLINK=-Wl,-dynamic
   SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
-  OBJS+=poll.o
-  ASTCFLAGS+=-DPOLLCOMPAT
+  # Mac on Intel CoreDuo does not need poll compatibility layer
+  ifneq ($(MARCH),i386)
+    OBJS+=poll.o
+    ASTCFLAGS+=-DPOLLCOMPAT
+  endif
 else
 #These are used for all but Darwin
   ASTLINK=-Wl,-E 

Modified: team/oej/siptransfer/README.siptransfer
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/README.siptransfer?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/README.siptransfer (original)
+++ team/oej/siptransfer/README.siptransfer Thu Apr  6 14:21:00 2006
@@ -25,3 +25,99 @@
 
 /Olle E. Johansson
 oej at edvina.net
+
+
+----------------DEVELOPMENT NOTES-------------------
+
+ * -- Transfer changes:
+ *	Parse SIP extensions supported by opposite side	-done
+ *	Announce that we support NOTIFY and replaces	-done
+ *	Use ${TRANSFER_CONTEXT} on transfers		-done
+ *	Removed 484 on REFER, a REFER is to a complete URI -done
+ *	Actually send replaces on invite if needed	-done
+ *	Add SIPTRANSFER variable to the new INVITE	-done
+ *	Send SIPDOMAIN to dialplan - will we find extension then? -done
+ *	Determine how to handle replace on refer...
+ * 	  - If the refer is to this server, replace (masq/rebridge) - done
+ *	  - If the refer is to another server -?
+ *		  If it is SIP, send INVITE with replace
+ *		  If it is not SIP, rebridge
+ * 	Make sure we have BRIDGEPEER	-done
+ *	Accept required: replaces	-done
+ * 	Fix SIP text message for call parking.	-done 
+ *	Fix remote one-legged transfers (voicemail etc)	-done
+ *	Fix remote early transfers (ringing/ring state)	-done
+ * -- Todo
+ *	Clean up debug messages 
+ *	Go over changes in channel.c again
+ *	Fix local one-legged transfers (ringing/ring state)	-done
+ *	Handle incoming 200 OK on NOTIFY correctly	-not done
+ *	   Transfer is done when we get 200 OK
+ *	   We should hangup only then
+ *	Send replaces only if supported		-not done
+ *	Parse incoming NOTIFY			-not done
+ *	Send NOTIFY until new call is up	-no solution yet, faking
+ *	Complete transfer (hangup call) only if we get 200 OK
+ *						-not done
+ *	Go back to call parking (transfer to 700 and find the problem)
+ *						-not done
+ *	Fix CANCEL on proxy auth required	-not done 
+ *	
+ * -- Tests
+ *      * Set up OUTBOUND call
+ *
+ *	Let the other end send a REFER to transfer to local extension
+ *	Let the other end send a REFER to transfer to remote SIP uri
+ *      Let the other end send a REFER to a not existing extension
+ *      Let the other end send a REFER to a busy extension
+ *
+ *      * Receive INBOUND CALL
+ *	Let the other end send a REFER to transfer to local extension
+ *      Handle the transfer
+ *      Let the other end send a REFER to a non existing extension
+ *      Let the other end send a REFER to a busy extension
+ *
+ *	exten => transfertest,1,answer
+ *	exten => transfertest,2,wait(2)
+ *	exten => transfertest,3,playback(beep)
+ *	exten => transfertest,4,transfer(SIP/localextension)
+ *	exten => transfertest,4,transfer(SIP/exten at remotehost)
+ *
+ *      * Receive INBOUND CALL
+ *	Send REFER to local extension
+ *	Send REFER to remote SIP uri
+ *
+ *	Test with devices 
+ *	-- Cisco		-- terry
+ *	-- Snom	360		-- olle
+ *	-- Sipura		-- olle
+ *	-- Xten Eye-Beam	-- olle
+ *	-- Polycom		-- olle/terry
+ *	-- GSX2000		-- olle
+ *
+ *	Test with another Asterisk 1.0
+ *	Test with another Asterisk with this patch	-- olle/terry
+ * 	
+ *	- Is the call coming back when transfer fails?
+ *			Blind transfer: yes
+ *			Attended transfer: no
+ *				-- not yet, not yet
+ * With Async GOTO we are messed up so we can't continue
+ * to use the channel
+ * - We need a new kind of bridge in Asterisk to do this
+ *   properly
+ * -------------------------------------------------------
+ * Authentication/security
+ *
+ *	With the allowtransfer setting (per peer or general)
+ *	you set who that we accept a incoming REFER from
+ *
+ *	Allowtransfer setting				Strict	Open	Closed
+ *	- Inbound call from authenticated user		yes	yes	no
+ *	- Inbound call from known authenticated peer	yes	yes	no
+ *	- Inbound call from known peer			no	yes	no
+ *	- Inbound call from unknown			no	yes	no
+ *	- Outbound call to known peer			yes	yes	no
+ *	- Outbound call to unknown			no	yes	no
+ *
+ */

Modified: team/oej/siptransfer/app.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/app.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/app.c (original)
+++ team/oej/siptransfer/app.c Thu Apr  6 14:21:00 2006
@@ -1157,7 +1157,7 @@
 		return AST_LOCK_FAILURE;
 	}
 
-	snprintf(fs, strlen(path) + 19, "%s/.lock-%08x", path, rand());
+	snprintf(fs, strlen(path) + 19, "%s/.lock-%08lx", path, ast_random());
 	fd = open(fs, O_WRONLY | O_CREAT | O_EXCL, 0600);
 	if (fd < 0) {
 		fprintf(stderr, "Unable to create lock file '%s': %s\n", path, strerror(errno));

Modified: team/oej/siptransfer/apps/app_alarmreceiver.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_alarmreceiver.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_alarmreceiver.c (original)
+++ team/oej/siptransfer/apps/app_alarmreceiver.c Thu Apr  6 14:21:00 2006
@@ -551,13 +551,12 @@
 
 		checksum = checksum % 15;
 
-		if(checksum){
+		if (checksum) {
 			database_increment("checksum-errors");
-			if(option_verbose >= 2){
+			if (option_verbose >= 2)
 				ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Nonzero checksum\n");
 			ast_log(LOG_DEBUG, "AlarmReceiver: Nonzero checksum\n");
 			continue;
-			}
 		}
 
 		/* Check the message type for correctness */

Modified: team/oej/siptransfer/apps/app_channelredirect.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_channelredirect.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_channelredirect.c (original)
+++ team/oej/siptransfer/apps/app_channelredirect.c Thu Apr  6 14:21:00 2006
@@ -99,15 +99,16 @@
 		context = NULL;
 	}
 
-	if (!(prio = ast_findlabel_extension(chan2, S_OR(context, chan2->context), S_OR(exten, chan2->exten),
-					     priority, chan2->cid.cid_num))) {
+	/* ast_findlabel_extension does not convert numeric priorities; it only does a lookup */
+	if (!(prio = atoi(priority)) && !(prio = ast_findlabel_extension(chan2, S_OR(context, chan2->context),
+									S_OR(exten, chan2->exten), priority, chan2->cid.cid_num))) {
 		ast_log(LOG_WARNING, "'%s' is not a known priority or label\n", priority);
 		goto chanquit;
 	}
 
-	ast_log(LOG_DEBUG, "Attempting async goto (%s) to %s\n", args.channel, args.label);
+	ast_log(LOG_DEBUG, "Attempting async goto (%s) to %s|%s|%d\n", args.channel, S_OR(context, chan2->context), S_OR(exten, chan2->exten), prio);
 
-	if (ast_async_goto_if_exists(chan2, context ? context : chan2->context, exten ? exten : chan2->exten, prio))
+	if (ast_async_goto_if_exists(chan2, S_OR(context, chan2->context), S_OR(exten, chan2->exten), prio))
 		ast_log(LOG_WARNING, "%s failed for %s\n", app, args.channel);
 	else
 		res = 0;

Modified: team/oej/siptransfer/apps/app_exec.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_exec.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_exec.c (original)
+++ team/oej/siptransfer/apps/app_exec.c Thu Apr  6 14:21:00 2006
@@ -2,8 +2,9 @@
  * Asterisk -- An open source telephony toolkit.
  *
  * Copyright (c) 2004 - 2005, Tilghman Lesher.  All rights reserved.
+ * Portions copyright (c) 2006, Philipp Dunkel.
  *
- * Tilghman Lesher <app_exec__v001 at the-tilghman.com>
+ * Tilghman Lesher <app_exec__v002 at the-tilghman.com>
  *
  * This code is released by the author with no restrictions on usage.
  *
@@ -19,7 +20,8 @@
  *
  * \brief Exec application
  *
- * \author Tilghman Lesher <app_exec__v001 at the-tilghman.com>
+ * \author Tilghman Lesher <app_exec__v002 at the-tilghman.com>
+ * \author Philipp Dunkel <philipp.dunkel at ebox.at>
  *
  * \ingroup applications
  */
@@ -43,18 +45,43 @@
 /* Maximum length of any variable */
 #define MAXRESULT	1024
 
-static char *tdesc = "Executes applications";
+static char *tdesc = "Executes dialplan applications";
+
+/*! Note
+ *
+ * The key difference between these two apps is exit status.  In a
+ * nutshell, Exec tries to be transparent as possible, behaving
+ * in exactly the same way as if the application it calls was
+ * directly invoked from the dialplan.
+ *
+ * TryExec, on the other hand, provides a way to execute applications
+ * and catch any possible fatal error without actually fatally
+ * affecting the dialplan.
+ */
 
 static char *app_exec = "Exec";
-
-static char *exec_synopsis = "Executes internal application";
-
+static char *exec_synopsis = "Executes dialplan application";
 static char *exec_descrip =
 "Usage: Exec(appname(arguments))\n"
 "  Allows an arbitrary application to be invoked even when not\n"
+"hardcoded into the dialplan.  If the underlying application\n"
+"terminates the dialplan, or if the application cannot be found,\n"
+"Exec will terminate the dialplan.\n"
+"  To invoke external applications, see the application System.\n"
+"  If you would like to catch any error instead, see TryExec.\n";
+
+static char *app_tryexec = "TryExec";
+static char *tryexec_synopsis = "Executes dialplan application, always returning";
+static char *tryexec_descrip =
+"Usage: TryExec(appname(arguments))\n"
+"  Allows an arbitrary application to be invoked even when not\n"
 "hardcoded into the dialplan. To invoke external applications\n"
-"see the application System. Returns whatever value the\n"
-"app returns or a non-zero value if the app cannot be found.\n";
+"see the application System.  Always returns to the dialplan.\n"
+"The channel variable TRYSTATUS will be set to:\n"
+"    SUCCESS   if the application returned zero\n"
+"    FAILED    if the application returned non-zero\n"
+"    NOAPP     if the application was not found or was not specified\n"
+"    NOMEMORY  if there was not enough memory to execute.\n";
 
 LOCAL_USER_DECL;
 
@@ -62,12 +89,10 @@
 {
 	int res=0;
 	struct localuser *u;
-	char *s, *appname, *endargs, args[MAXRESULT];
+	char *s, *appname, *endargs, args[MAXRESULT] = "";
 	struct ast_app *app;
 
 	LOCAL_USER_ADD(u);
-
-	memset(args, 0, MAXRESULT);
 
 	/* Check and parse arguments */
 	if (data) {
@@ -96,11 +121,51 @@
 	return res;
 }
 
+static int tryexec_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	char *s, *appname, *endargs, args[MAXRESULT] = "";
+	struct ast_app *app;
+
+	LOCAL_USER_ADD(u);
+
+	/* Check and parse arguments */
+	if (data) {
+		if ((s = ast_strdupa(data))) {
+			appname = strsep(&s, "(");
+			if (s) {
+				endargs = strrchr(s, ')');
+				if (endargs)
+					*endargs = '\0';
+				pbx_substitute_variables_helper(chan, s, args, MAXRESULT - 1);
+			}
+			if (appname) {
+				app = pbx_findapp(appname);
+				if (app) {
+					res = pbx_exec(chan, app, args);
+					pbx_builtin_setvar_helper(chan, "TRYSTATUS", res ? "FAILED" : "SUCCESS");
+				} else {
+					ast_log(LOG_WARNING, "Could not find application (%s)\n", appname);
+					pbx_builtin_setvar_helper(chan, "TRYSTATUS", "NOAPP");
+				}
+			}
+		} else {
+			ast_log(LOG_ERROR, "Out of memory\n");
+			pbx_builtin_setvar_helper(chan, "TRYSTATUS", "NOMEMORY");
+		}
+	}
+
+	LOCAL_USER_REMOVE(u);
+	return 0;
+}
+
 int unload_module(void)
 {
 	int res;
 
 	res = ast_unregister_application(app_exec);
+	res |= ast_unregister_application(app_tryexec);
 
 	STANDARD_HANGUP_LOCALUSERS;
 
@@ -109,7 +174,9 @@
 
 int load_module(void)
 {
-	return ast_register_application(app_exec, exec_exec, exec_synopsis, exec_descrip);
+	int res = ast_register_application(app_exec, exec_exec, exec_synopsis, exec_descrip);
+	res |= ast_register_application(app_tryexec, tryexec_exec, tryexec_synopsis, tryexec_descrip);
+	return res;
 }
 
 char *description(void)

Modified: team/oej/siptransfer/apps/app_page.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_page.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_page.c (original)
+++ team/oej/siptransfer/apps/app_page.c Thu Apr  6 14:21:00 2006
@@ -143,7 +143,7 @@
 	char *tech, *resource;
 	char meetmeopts[80];
 	struct ast_flags flags = { 0 };
-	unsigned int confid = rand();
+	unsigned int confid = ast_random();
 	struct ast_app *app;
 	char *tmp;
 	int res=0;

Modified: team/oej/siptransfer/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_queue.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_queue.c (original)
+++ team/oej/siptransfer/apps/app_queue.c Thu Apr  6 14:21:00 2006
@@ -1211,7 +1211,7 @@
 
 	ast_mutex_lock(&qe->parent->lock);
 	if (newvalue <= qe->parent->servicelevel)
-       		qe->parent->callscompletedinsl++;
+		qe->parent->callscompletedinsl++;
 	oldvalue = qe->parent->holdtime;
 	qe->parent->holdtime = (((oldvalue << 2) - oldvalue) + newvalue) >> 2;
 	ast_mutex_unlock(&qe->parent->lock);
@@ -1753,7 +1753,7 @@
 				if (f) {
 					if (f->frametype == AST_FRAME_CONTROL) {
 						switch(f->subclass) {
-		    			case AST_CONTROL_ANSWER:
+						case AST_CONTROL_ANSWER:
 							/* This is our guy if someone answered. */
 							if (!peer) {
 								if (option_verbose > 2)
@@ -1992,7 +1992,7 @@
 		tmp->metric += mem->penalty * 1000000;
 		break;
 	case QUEUE_STRATEGY_RANDOM:
-		tmp->metric = rand() % 1000;
+		tmp->metric = ast_random() % 1000;
 		tmp->metric += mem->penalty * 1000000;
 		break;
 	case QUEUE_STRATEGY_FEWESTCALLS:
@@ -2193,7 +2193,7 @@
 				/* Agent must have hung up */
 				ast_log(LOG_WARNING, "Agent on %s hungup on the customer.  They're going to be pissed.\n", peer->name);
 				ast_queue_log(queuename, qe->chan->uniqueid, peer->name, "AGENTDUMP", "%s", "");
-                                record_abandoned(qe);
+				record_abandoned(qe);
 				if (qe->parent->eventwhencalled) {
 					manager_event(EVENT_FLAG_AGENT, "AgentDump",
 						      "Queue: %s\r\n"
@@ -2223,7 +2223,7 @@
 		if (res < 0) {
 			ast_queue_log(queuename, qe->chan->uniqueid, peer->name, "SYSCOMPAT", "%s", "");
 			ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", qe->chan->name, peer->name);
-                        record_abandoned(qe);
+		record_abandoned(qe);
 			ast_hangup(peer);
 			return -1;
 		}
@@ -2241,7 +2241,7 @@
 			else {
 				/* Last ditch effort -- no CDR, make up something */
 				char tmpid[256];
-				snprintf(tmpid, sizeof(tmpid), "chan-%x", rand());
+				snprintf(tmpid, sizeof(tmpid), "chan-%lx", ast_random());
 				ast_monitor_start(which, qe->parent->monfmt, tmpid, 1 );
 			}
 			if (qe->parent->monjoin)
@@ -2403,7 +2403,7 @@
 				free(last_member);
 
 				if (queue_persistent_members)
-				    dump_queue_members(q);
+					dump_queue_members(q);
 
 				res = RES_OKAY;
 			} else {
@@ -2440,17 +2440,17 @@
 				new_member->next = q->members;
 				q->members = new_member;
 				manager_event(EVENT_FLAG_AGENT, "QueueMemberAdded",
-					"Queue: %s\r\n"
-					"Location: %s\r\n"
-					"Membership: %s\r\n"
-					"Penalty: %d\r\n"
-					"CallsTaken: %d\r\n"
-					"LastCall: %d\r\n"
-					"Status: %d\r\n"
-					"Paused: %d\r\n",
-				q->name, new_member->interface, new_member->dynamic ? "dynamic" : "static",
-				new_member->penalty, new_member->calls, (int)new_member->lastcall, new_member->status, new_member->paused);
-					
+						"Queue: %s\r\n"
+						"Location: %s\r\n"
+						"Membership: %s\r\n"
+						"Penalty: %d\r\n"
+						"CallsTaken: %d\r\n"
+						"LastCall: %d\r\n"
+						"Status: %d\r\n"
+						"Paused: %d\r\n",
+						q->name, new_member->interface, new_member->dynamic ? "dynamic" : "static",
+						new_member->penalty, new_member->calls, (int)new_member->lastcall, new_member->status, new_member->paused);
+
 				if (dump)
 					dump_queue_members(q);
 
@@ -2489,7 +2489,7 @@
 				mem->paused = paused;
 
 				if (queue_persistent_members)
-				    dump_queue_members(q);
+					dump_queue_members(q);
 
 				ast_queue_log(q->name, "NONE", interface, (paused ? "PAUSE" : "UNPAUSE"), "%s", "");
 
@@ -2944,7 +2944,7 @@
 check_turns:
 		if (ringing) {
 			ast_indicate(chan, AST_CONTROL_RINGING);
-		} else {              
+		} else {
 			ast_moh_start(chan, qe.moh);
 		}
 		for (;;) {
@@ -2958,8 +2958,8 @@
 				 ast_queue_log(args.queuename, chan->uniqueid, "NONE", "ABANDON", "%d|%d|%ld", qe.pos, qe.opos, (long)time(NULL) - qe.start);
 				if (option_verbose > 2) {
 					ast_verbose(VERBOSE_PREFIX_3 "User disconnected from queue %s while waiting their turn\n", args.queuename);
-					res = -1;
 				}
+				res = -1;
 				break;
 			}
 			if (!res) 
@@ -2981,7 +2981,7 @@
 
 				/* Leave if we have exceeded our queuetimeout */
 				if (qe.expire && (time(NULL) > qe.expire)) {
-                                        record_abandoned(&qe);
+					record_abandoned(&qe);
 					reason = QUEUE_TIMEOUT;
 					res = 0;
 					ast_queue_log(args.queuename, chan->uniqueid,"NONE", "EXITWITHTIMEOUT", "%d", qe.pos);
@@ -3057,8 +3057,8 @@
 					ast_queue_log(args.queuename, chan->uniqueid, "NONE", "ABANDON", "%d|%d|%ld", qe.pos, qe.opos, (long)time(NULL) - qe.start);
 					if (option_verbose > 2) {
 						ast_verbose(VERBOSE_PREFIX_3 "User disconnected from queue %s when they almost made it\n", args.queuename);
-						res = -1;
 					}
+					res = -1;
 					break;
 				}
 				if (res && valid_exit(&qe, res)) {
@@ -3067,12 +3067,10 @@
 				}
 				/* exit after 'timeout' cycle if 'n' option enabled */
 				if (go_on) {
-					if (option_verbose > 2) {
+					if (option_verbose > 2)
 						ast_verbose(VERBOSE_PREFIX_3 "Exiting on time-out cycle\n");
-						res = -1;
-					}
 					ast_queue_log(args.queuename, chan->uniqueid, "NONE", "EXITWITHTIMEOUT", "%d", qe.pos);
-                                        record_abandoned(&qe);
+					record_abandoned(&qe);
 					reason = QUEUE_TIMEOUT;
 					res = 0;
 					break;

Modified: team/oej/siptransfer/apps/app_senddtmf.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_senddtmf.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_senddtmf.c (original)
+++ team/oej/siptransfer/apps/app_senddtmf.c Thu Apr  6 14:21:00 2006
@@ -52,7 +52,7 @@
 
 static char *descrip = 
 " SendDTMF(digits[|timeout_ms]): Sends DTMF digits on a channel. \n"
-" Accepted digits: 0-9, *#abcd\n"
+" Accepted digits: 0-9, *#abcd, w (.5s pause)\n"
 " The application will either pass the assigned digits or terminate if it\n"
 " encounters an error.\n";
 

Modified: team/oej/siptransfer/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_voicemail.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_voicemail.c (original)
+++ team/oej/siptransfer/apps/app_voicemail.c Thu Apr  6 14:21:00 2006
@@ -1753,11 +1753,11 @@
 			fprintf(p, "Subject: New message %d in mailbox %s\n", msgnum + 1, mailbox);
 		else
 			fprintf(p, "Subject: [PBX]: New message %d in mailbox %s\n", msgnum + 1, mailbox);
-		fprintf(p, "Message-ID: <Asterisk-%d-%d-%s-%d@%s>\n", msgnum, (unsigned int)rand(), mailbox, getpid(), host);
+		fprintf(p, "Message-ID: <Asterisk-%d-%d-%s-%d@%s>\n", msgnum, (unsigned int)ast_random(), mailbox, getpid(), host);
 		fprintf(p, "MIME-Version: 1.0\n");
 		if (attach_user_voicemail) {
 			/* Something unique. */
-			snprintf(bound, sizeof(bound), "voicemail_%d%s%d%d", msgnum, mailbox, getpid(), (unsigned int)rand());
+			snprintf(bound, sizeof(bound), "voicemail_%d%s%d%d", msgnum, mailbox, getpid(), (unsigned int)ast_random());
 
 			fprintf(p, "Content-Type: multipart/mixed; boundary=\"%s\"\n\n\n", bound);
 

Modified: team/oej/siptransfer/channel.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channel.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/channel.c (original)
+++ team/oej/siptransfer/channel.c Thu Apr  6 14:21:00 2006
@@ -333,7 +333,7 @@
 	shutting_down = 1;
 	if (hangup) {
 		AST_LIST_LOCK(&channels);
-		AST_LIST_TRAVERSE(&channels, c, list)
+		AST_LIST_TRAVERSE(&channels, c, chan_list)
 			ast_softhangup(c, AST_SOFTHANGUP_SHUTDOWN);
 		AST_LIST_UNLOCK(&channels);
 	}
@@ -345,7 +345,7 @@
 	struct ast_channel *c;
 	int cnt = 0;
 	AST_LIST_LOCK(&channels);
-	AST_LIST_TRAVERSE(&channels, c, list)
+	AST_LIST_TRAVERSE(&channels, c, chan_list)
 		cnt++;
 	AST_LIST_UNLOCK(&channels);
 	return cnt;
@@ -682,7 +682,7 @@
 	tmp->tech = &null_tech;
 
 	AST_LIST_LOCK(&channels);
-	AST_LIST_INSERT_HEAD(&channels, tmp, list);
+	AST_LIST_INSERT_HEAD(&channels, tmp, chan_list);
 	AST_LIST_UNLOCK(&channels);
 	return tmp;
 }
@@ -816,7 +816,7 @@
 
 	for (retries = 0; retries < 10; retries++) {
 		AST_LIST_LOCK(&channels);
-		AST_LIST_TRAVERSE(&channels, c, list) {
+		AST_LIST_TRAVERSE(&channels, c, chan_list) {
 			if (!prev) {
 				/* want head of list */
 				if (!name && !exten)
@@ -845,7 +845,7 @@
 						break;
 				}
 			} else if (c == prev) { /* found, return c->next */
-				c = AST_LIST_NEXT(c, list);
+				c = AST_LIST_NEXT(c, chan_list);
 				break;
 			}
 		}
@@ -952,7 +952,7 @@
 	headp=&chan->varshead;
 	
 	AST_LIST_LOCK(&channels);
-	AST_LIST_REMOVE(&channels, chan, list);
+	AST_LIST_REMOVE(&channels, chan, chan_list);
 	/* Lock and unlock the channel just to be sure nobody
 	   has it locked still */
 	ast_mutex_lock(&chan->lock);

Modified: team/oej/siptransfer/channels/chan_agent.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_agent.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_agent.c (original)
+++ team/oej/siptransfer/channels/chan_agent.c Thu Apr  6 14:21:00 2006
@@ -924,7 +924,7 @@
 			tmp->rawreadformat = AST_FORMAT_SLINEAR;
 		}
 		if (p->pending)
-			ast_string_field_build(tmp, name, "Agent/P%s-%d", p->agent, rand() & 0xffff);
+			ast_string_field_build(tmp, name, "Agent/P%s-%d", p->agent, ast_random() & 0xffff);
 		else
 			ast_string_field_build(tmp, name, "Agent/%s", p->agent);
 		/* Safe, agentlock already held */

Modified: team/oej/siptransfer/channels/chan_h323.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_h323.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_h323.c (original)
+++ team/oej/siptransfer/channels/chan_h323.c Thu Apr  6 14:21:00 2006
@@ -78,6 +78,7 @@
 #include "asterisk/cli.h"
 #include "asterisk/dsp.h"
 #include "asterisk/causes.h"
+#include "asterisk/stringfields.h"
 #ifdef __cplusplus
 }
 #endif
@@ -101,12 +102,13 @@
 int h323debug;
 
 /** Variables required by Asterisk */
-static const char type[] = "H323";
 static const char desc[] = "The NuFone Network's Open H.323 Channel Driver";
 static const char tdesc[] = "The NuFone Network's Open H.323 Channel Driver";
 static const char config[] = "h323.conf";
 static char default_context[AST_MAX_CONTEXT] = "default";
 static struct sockaddr_in bindaddr;
+
+#define GLOBAL_CAPABILITY (AST_FORMAT_G723_1 | AST_FORMAT_GSM | AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_G729A | AST_FORMAT_H261)
 
 /** H.323 configuration values */
 static int h323_signalling_port = 1720;
@@ -206,9 +208,9 @@
 static int oh323_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
 
 static const struct ast_channel_tech oh323_tech = {
-	.type = type,
+	.type = "H323",
 	.description = tdesc,
-	.capabilities = AST_FORMAT_ULAW,
+	.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
 	.properties = AST_CHAN_TP_WANTSJITTER,
 	.requester = oh323_request,
 	.send_digit = oh323_digit,
@@ -496,7 +498,7 @@
 	if (c->hangupcause) {
 		q931cause = c->hangupcause;
 	} else {
-		char *cause = pbx_builtin_getvar_helper(c, "DIALSTATUS");
+		const char *cause = pbx_builtin_getvar_helper(c, "DIALSTATUS");
 		if (cause) {
 			if (!strcmp(cause, "CONGESTION")) {
 				q931cause = AST_CAUSE_NORMAL_CIRCUIT_CONGESTION;
@@ -735,14 +737,13 @@
 	ast_mutex_lock(&pvt->lock);
 	if (ch) {
 		ch->tech = &oh323_tech;
-		snprintf(ch->name, sizeof(ch->name), "H323/%s", host);
+		ast_string_field_build(ch, name, "H323/%s", host);
 		ch->nativeformats = pvt->options.capability;
 		if (!ch->nativeformats) {
 			ch->nativeformats = global_options.capability;
 		}
 		pvt->nativeformats = ch->nativeformats;
 		fmt = ast_best_codec(ch->nativeformats);
-		ch->type = type;
 		ch->fds[0] = ast_rtp_fd(pvt->rtp);
 		if (state == AST_STATE_RING) {
 			ch->rings = 1;
@@ -765,7 +766,7 @@
 		strncpy(ch->exten, pvt->exten, sizeof(ch->exten) - 1);		
 		ch->priority = 1;
 		if (!ast_strlen_zero(pvt->accountcode)) {
-			strncpy(ch->accountcode, pvt->accountcode, sizeof(ch->accountcode) - 1);
+			ast_string_field_set(ch, accountcode, pvt->accountcode);
 		}
 		if (pvt->amaflags) {
 			ch->amaflags = pvt->amaflags;
@@ -1234,7 +1235,6 @@
   */
 void connection_made(unsigned call_reference, const char *token)
 {
-	struct ast_channel *c = NULL;
 	struct oh323_pvt *pvt;
 
 	if (h323debug)
@@ -1436,7 +1436,6 @@
   */
 void chan_ringing(unsigned call_reference, const char *token)
 {
-	struct ast_channel *c = NULL;
 	struct oh323_pvt *pvt;
 
 	if (h323debug)
@@ -1479,7 +1478,7 @@
 			break;
 #if 1
 #ifdef DEBUG_THREADS
-		ast_log(LOG_NOTICE, "Avoiding H.323 destory deadlock on %s, locked at %ld/%d by %s (%s:%d)\n", call_token, pvt->owner->lock.thread, pvt->owner->lock.reentrancy, pvt->owner->lock.func, pvt->owner->lock.file, pvt->owner->lock.lineno);
+		ast_log(LOG_NOTICE, "Avoiding H.323 destory deadlock on %s, locked at %ld/%d by %s (%s:%d)\n", call_token, pvt->owner->lock.thread[0], pvt->owner->lock.reentrancy, pvt->owner->lock.func[0], pvt->owner->lock.file[0], pvt->owner->lock.lineno[0]);
 #else
 		ast_log(LOG_NOTICE, "Avoiding H.323 destory deadlock on %s\n", call_token);
 #endif
@@ -2026,7 +2025,7 @@
 	memset(&global_options, 0, sizeof(global_options));
 	global_options.dtmfcodec = 101;
 	global_options.dtmfmode = H323_DTMF_RFC2833;
-	global_options.capability = ~0;	/* All capabilities */
+	global_options.capability = GLOBAL_CAPABILITY;
 	global_options.bridge = 1;		/* Do native bridging by default */
 	v = ast_variable_browse(cfg, "general");
 	while(v) {
@@ -2282,7 +2281,7 @@
 	}
 }
 
-static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
+static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
 {
 	/* XXX Deal with Video */
 	struct oh323_pvt *pvt;
@@ -2308,7 +2307,7 @@
 }
 
 static struct ast_rtp_protocol oh323_rtp = {
-	.type = type,
+	.type = "H323",
 	.get_rtp_info = oh323_get_rtp_peer,
 	.get_vrtp_info = oh323_get_vrtp_peer,
 	.set_rtp_peer=  oh323_set_rtp_peer,
@@ -2334,7 +2333,7 @@
 	} else {
 		/* Make sure we can register our channel type */
 		if (ast_channel_register(&oh323_tech)) {
-			ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
+			ast_log(LOG_ERROR, "Unable to register channel class 'H323'\n");
 			h323_end_process();
 			return -1;
 		}

Modified: team/oej/siptransfer/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_iax2.c?rev=17987&r1=17986&r2=17987&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_iax2.c (original)
+++ team/oej/siptransfer/channels/chan_iax2.c Thu Apr  6 14:21:00 2006
@@ -778,6 +778,7 @@
 static int send_command_immediate(struct chan_iax2_pvt *, char, int, unsigned int, const unsigned char *, int, int);
 static int send_command_final(struct chan_iax2_pvt *, char, int, unsigned int, const unsigned char *, int, int);
 static int send_command_transfer(struct chan_iax2_pvt *, char, int, unsigned int, const unsigned char *, int);
+static struct iax2_peer *build_peer(const char *name, struct ast_variable *v, int temponly);
 static struct iax2_user *build_user(const char *name, struct ast_variable *v, int temponly);
 static void destroy_user(struct iax2_user *user);
 static int expire_registry(void *data);
@@ -1291,7 +1292,7 @@
 		last++;
 	else
 		last = s;
-	snprintf(s2, strlen(s) + 100, "/var/tmp/%s-%ld", last, (unsigned long)rand());
+	snprintf(s2, strlen(s) + 100, "/var/tmp/%s-%ld", last, (unsigned long)ast_random());
 	res = stat(s, &stbuf);
 	if (res < 0) {
 		ast_log(LOG_WARNING, "Failed to stat '%s': %s\n", s, strerror(errno));
@@ -1609,19 +1610,14 @@
 	int res;
 	char iabuf[INET_ADDRSTRLEN];
 	int callno = f->callno;
+
+	/* Don't send if there was an error, but return error instead */
+	if (!callno || !iaxs[callno] || iaxs[callno]->error)
+	    return -1;
 	
 	/* Called with iaxsl held */
-	if (!iaxs[callno])
-		return -1;
 	if (option_debug > 2 && iaxdebug)
 		ast_log(LOG_DEBUG, "Sending %d on %d/%d to %s:%d\n", f->ts, callno, iaxs[callno]->peercallno, ast_inet_ntoa(iabuf, sizeof(iabuf), iaxs[callno]->addr.sin_addr), ntohs(iaxs[callno]->addr.sin_port));
-	/* Don't send if there was an error, but return error instead */
-	if (!callno) {
-		ast_log(LOG_WARNING, "Call number = %d\n", callno);
-		return -1;
-	}
-	if (iaxs[callno]->error)
-		return -1;
 	if (f->transfer) {
 		if (iaxdebug)
 			iax_showframe(f, NULL, 0, &iaxs[callno]->transfer, f->datalen - sizeof(struct ast_iax2_full_hdr));
@@ -1642,6 +1638,30 @@
 	return res;
 }
 
+static void iax2_destroy_helper(struct chan_iax2_pvt *pvt)
+{
+	/* No more pings or lagrq's */
+	if (pvt->pingid > -1)
+		ast_sched_del(sched, pvt->pingid);
+	pvt->pingid = -1;
+	if (pvt->lagid > -1)
+		ast_sched_del(sched, pvt->lagid);
+	pvt->lagid = -1;
+	if (pvt->autoid > -1)
+		ast_sched_del(sched, pvt->autoid);
+	pvt->autoid = -1;
+	if (pvt->authid > -1)
+		ast_sched_del(sched, pvt->authid);
+	pvt->authid = -1;
+	if (pvt->initid > -1)
+		ast_sched_del(sched, pvt->initid);
+	pvt->initid = -1;
+#ifdef NEWJB
+	if (pvt->jbid > -1)
+		ast_sched_del(sched, pvt->jbid);
+	pvt->jbid = -1;
+#endif
+}
 
 static int iax2_predestroy(int callno)
 {
@@ -1654,27 +1674,7 @@
 		return -1;
 	}
 	if (!ast_test_flag(pvt, IAX_ALREADYGONE)) {
-		/* No more pings or lagrq's */
-		if (pvt->pingid > -1)
-			ast_sched_del(sched, pvt->pingid);
-		if (pvt->lagid > -1)
-			ast_sched_del(sched, pvt->lagid);
-		if (pvt->autoid > -1)
-			ast_sched_del(sched, pvt->autoid);
-		if (pvt->authid > -1)
-			ast_sched_del(sched, pvt->authid);
-		if (pvt->initid > -1)
-			ast_sched_del(sched, pvt->initid);
-#ifdef NEWJB
-		if (pvt->jbid > -1)
-			ast_sched_del(sched, pvt->jbid);
-		pvt->jbid = -1;
-#endif
-		pvt->pingid = -1;
-		pvt->lagid = -1;
-		pvt->autoid = -1;
-		pvt->initid = -1;
-		pvt->authid = -1;
+		iax2_destroy_helper(pvt);
 		ast_set_flag(pvt, IAX_ALREADYGONE);	
 	}
 	c = pvt->owner;
@@ -1729,27 +1729,7 @@
 	if (pvt) {
 		if (!owner)
 			pvt->owner = NULL;
-		/* No more pings or lagrq's */
-		if (pvt->pingid > -1)
-			ast_sched_del(sched, pvt->pingid);
-		if (pvt->lagid > -1)
-			ast_sched_del(sched, pvt->lagid);
-		if (pvt->autoid > -1)
-			ast_sched_del(sched, pvt->autoid);
-		if (pvt->authid > -1)
-			ast_sched_del(sched, pvt->authid);
-		if (pvt->initid > -1)
-			ast_sched_del(sched, pvt->initid);
-#ifdef NEWJB
-		if (pvt->jbid > -1)
-			ast_sched_del(sched, pvt->jbid);
-		pvt->jbid = -1;
-#endif
-		pvt->pingid = -1;
-		pvt->lagid = -1;
-		pvt->autoid = -1;
-		pvt->authid = -1;
-		pvt->initid = -1;
+		iax2_destroy_helper(pvt);
 		if (pvt->bridgetrans)
 			ast_translator_free_path(pvt->bridgetrans);
 		pvt->bridgetrans = NULL;
@@ -2706,12 +2686,6 @@
 	return 0;
 }
 
-static struct iax2_peer *build_peer(const char *name, struct ast_variable *v, int temponly);
-static struct iax2_user *build_user(const char *name, struct ast_variable *v, int temponly);
-
-static void destroy_user(struct iax2_user *user);
-static int expire_registry(void *data);
-
 static struct iax2_peer *realtime_peer(const char *peername, struct sockaddr_in *sin)
 {
 	struct ast_variable *var;
@@ -3275,7 +3249,7 @@
 	int res;
 	struct iax_ie_data ied0;
 	struct iax_ie_data ied1;
-	unsigned int transferid = rand();
+	unsigned int transferid = (unsigned int)ast_random();
 	memset(&ied0, 0, sizeof(ied0));
 	iax_ie_append_addr(&ied0, IAX_IE_APPARENT_ADDR, &iaxs[callno1]->addr);
 	iax_ie_append_short(&ied0, IAX_IE_CALLNO, iaxs[callno1]->peercallno);
@@ -3816,9 +3790,9 @@
 	ms = ast_tvdiff_ms(ast_tvnow(), p->rxcore);
 #ifdef IAXTESTS
 	if (test_jit) {
-		if (!test_jitpct || ((100.0 * rand() / (RAND_MAX + 1.0)) < test_jitpct)) {
-			jit = (int)((float)test_jit * rand() / (RAND_MAX + 1.0));
-			if ((int)(2.0 * rand() / (RAND_MAX + 1.0)))
+		if (!test_jitpct || ((100.0 * ast_random() / (RAND_MAX + 1.0)) < test_jitpct)) {
+			jit = (int)((float)test_jit * ast_random() / (RAND_MAX + 1.0));
+			if ((int)(2.0 * ast_random() / (RAND_MAX + 1.0)))
 				jit = -jit;
 			ms += jit;
 		}
@@ -5170,7 +5144,7 @@
 	memset(&ied, 0, sizeof(ied));
 	iax_ie_append_short(&ied, IAX_IE_AUTHMETHODS, p->authmethods);
 	if (p->authmethods & (IAX_AUTH_MD5 | IAX_AUTH_RSA)) {
-		snprintf(p->challenge, sizeof(p->challenge), "%d", rand());
+		snprintf(p->challenge, sizeof(p->challenge), "%d", (int)ast_random());
 		iax_ie_append_str(&ied, IAX_IE_CHALLENGE, p->challenge);
 	}
 	if (p->encmethods)
@@ -5720,17 +5694,19 @@
 	if (reg->expire > -1)
 		ast_sched_del(sched, reg->expire);
 	reg->expire = ast_sched_add(sched, (5 * reg->refresh / 6) * 1000, iax2_do_register_s, reg);
-	if ((inaddrcmp(&oldus, &reg->us) || (reg->messages != oldmsgs)) && (option_verbose > 2)) {
-		if (reg->messages > 65534)
-			snprintf(msgstatus, sizeof(msgstatus), " with message(s) waiting\n");
-		else if (reg->messages > 1)
-			snprintf(msgstatus, sizeof(msgstatus), " with %d messages waiting\n", reg->messages);
-		else if (reg->messages > 0)

[... 9468 lines stripped ...]


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