[asterisk-commits] branch oej/astum - r8139 in /team/oej/astum: ./ apps/ channels/ include/aster...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Tue Jan 17 12:14:30 MST 2006


Author: oej
Date: Tue Jan 17 13:14:24 2006
New Revision: 8139

URL: http://svn.digium.com/view/asterisk?rev=8139&view=rev
Log:
Merged revisions 8113-8114,8118,8120,8123,8125,8127,8133,8135-8138 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r8113 | kpfleming | 2006-01-17 00:52:02 +0100 (Tue, 17 Jan 2006) | 2 lines

block this revision, fix is different in this branch

................
r8114 | kpfleming | 2006-01-17 00:52:51 +0100 (Tue, 17 Jan 2006) | 2 lines

check rlimit _after_ reading config file, so that if 'dumpcore' is specified there it will take effect

................
r8118 | oej | 2006-01-17 04:05:43 +0100 (Tue, 17 Jan 2006) | 2 lines

Doxygen update

................
r8120 | kpfleming | 2006-01-17 06:15:33 +0100 (Tue, 17 Jan 2006) | 2 lines

don't generate any message for native bridge attempts unless all the basic checks have passed and we're actually going to try it

................
r8123 | kpfleming | 2006-01-17 14:13:31 +0100 (Tue, 17 Jan 2006) | 10 lines

Merged revisions 8122 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r8122 | kpfleming | 2006-01-17 07:11:55 -0600 (Tue, 17 Jan 2006) | 2 lines

update CLI copyright notice

........

................
r8125 | mogorman | 2006-01-17 17:58:09 +0100 (Tue, 17 Jan 2006) | 10 lines

Merged revisions 7963 via svnmerge from 
https://svn.digium.com/svn/asterisk/branches/1.2

........
r7963 | mogorman | 2006-01-10 22:38:07 -0600 (Tue, 10 Jan 2006) | 2 lines

Minor typo refrenced in 6191

........

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r8127 | mogorman | 2006-01-17 18:25:53 +0100 (Tue, 17 Jan 2006) | 2 lines

Added tab completion for help.  bug 6074

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r8133 | mattf | 2006-01-17 19:20:33 +0100 (Tue, 17 Jan 2006) | 2 lines

Check to see if arg is NULL before passing (#6094)

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r8135 | bweschke | 2006-01-17 19:31:03 +0100 (Tue, 17 Jan 2006) | 3 lines

 Fix compiler warning.


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r8136 | oej | 2006-01-17 19:54:56 +0100 (Tue, 17 Jan 2006) | 3 lines

- Logging clean up
- Whitespace removed and added, formatting fixed

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r8137 | oej | 2006-01-17 19:56:57 +0100 (Tue, 17 Jan 2006) | 2 lines

Extra comma causing compilation errors...

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r8138 | oej | 2006-01-17 20:03:04 +0100 (Tue, 17 Jan 2006) | 2 lines

Portability - compilation warning on Mac OS/X removed

................

Modified:
    team/oej/astum/   (props changed)
    team/oej/astum/apps/app_dial.c
    team/oej/astum/asterisk.c
    team/oej/astum/channel.c
    team/oej/astum/channels/chan_local.c
    team/oej/astum/channels/chan_misdn.c
    team/oej/astum/channels/chan_vpb.c
    team/oej/astum/channels/chan_zap.c
    team/oej/astum/cli.c
    team/oej/astum/include/asterisk/doxyref.h
    team/oej/astum/logger.c
    team/oej/astum/manager.c
    team/oej/astum/res/res_features.c
    team/oej/astum/rtp.c

Propchange: team/oej/astum/
------------------------------------------------------------------------------
--- svnmerge-blocked (original)
+++ svnmerge-blocked Tue Jan 17 13:14:24 2006
@@ -1,1 +1,1 @@
-/branches/1.2:7490,7497,7517,7529,7546,7550,7552,7557,7580,7586,7595,7605,7641,7663,7706,7738,7771,7792,7812,7870-7871,7898-7900,7915,7960,7965,7970,7976,8047
+/branches/1.2:7490,7497,7517,7529,7546,7550,7552,7557,7580,7586,7595,7605,7641,7663,7706,7738,7771,7792,7812,7870-7871,7898-7900,7915,7960,7965,7970,7976,8047,8112

Propchange: team/oej/astum/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jan 17 13:14:24 2006
@@ -1,1 +1,1 @@
-/trunk:1-8105
+/trunk:1-8138

Modified: team/oej/astum/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/apps/app_dial.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/apps/app_dial.c (original)
+++ team/oej/astum/apps/app_dial.c Tue Jan 17 13:14:24 2006
@@ -189,30 +189,30 @@
 "to the Dial application.\n";
 
 enum {
-	OPT_ANNOUNCE = (1 << 0),
-	OPT_RESETCDR = (1 << 1),
-	OPT_DTMF_EXIT = (1 << 2),
-	OPT_SENDDTMF = (1 << 3),
-	OPT_FORCECLID = (1 << 4),
-	OPT_GO_ON = (1 << 5),
-	OPT_CALLEE_HANGUP = (1 << 6),
-	OPT_CALLER_HANGUP = (1 << 7),
-	OPT_PRIORITY_JUMP = (1 << 8),
-	OPT_DURATION_LIMIT = (1 << 9),
-	OPT_MUSICBACK = (1 << 10),
-	OPT_CALLEE_MACRO = (1 << 11),
-	OPT_SCREEN_NOINTRO = (1 << 12),
-	OPT_SCREEN_NOCLID = (1 << 13),
-	OPT_ORIGINAL_CLID = (1 << 14),
-	OPT_SCREENING = (1 << 15),
-	OPT_PRIVACY = (1 << 16),
-	OPT_RINGBACK = (1 << 17),
-	OPT_DURATION_STOP = (1 << 18),
-	OPT_CALLEE_TRANSFER = (1 << 19),
-	OPT_CALLER_TRANSFER = (1 << 20),
-	OPT_CALLEE_MONITOR = (1 << 21),
-	OPT_CALLER_MONITOR = (1 << 22),
-	OPT_GOTO = (1 << 23),
+	OPT_ANNOUNCE =		(1 << 0),
+	OPT_RESETCDR =		(1 << 1),
+	OPT_DTMF_EXIT =		(1 << 2),
+	OPT_SENDDTMF =		(1 << 3),
+	OPT_FORCECLID =		(1 << 4),
+	OPT_GO_ON =		(1 << 5),
+	OPT_CALLEE_HANGUP =	(1 << 6),
+	OPT_CALLER_HANGUP =	(1 << 7),
+	OPT_PRIORITY_JUMP =	(1 << 8),
+	OPT_DURATION_LIMIT =	(1 << 9),
+	OPT_MUSICBACK =		(1 << 10),
+	OPT_CALLEE_MACRO =	(1 << 11),
+	OPT_SCREEN_NOINTRO =	(1 << 12),
+	OPT_SCREEN_NOCLID =	(1 << 13),
+	OPT_ORIGINAL_CLID =	(1 << 14),
+	OPT_SCREENING =		(1 << 15),
+	OPT_PRIVACY =		(1 << 16),
+	OPT_RINGBACK =		(1 << 17),
+	OPT_DURATION_STOP =	(1 << 18),
+	OPT_CALLEE_TRANSFER =	(1 << 19),
+	OPT_CALLER_TRANSFER =	(1 << 20),
+	OPT_CALLEE_MONITOR =	(1 << 21),
+	OPT_CALLER_MONITOR =	(1 << 22),
+	OPT_GOTO =		(1 << 23),
 } dial_exec_option_flags;
 
 #define DIAL_STILLGOING			(1 << 30)
@@ -425,7 +425,7 @@
 					ast_goto_if_exists(in, in->context, in->exten, in->priority + 101);
 			} else {
 				if (option_verbose > 2)
-					ast_verbose( VERBOSE_PREFIX_2 "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, numbusy, numcongestion, numnochan);
+					ast_verbose(VERBOSE_PREFIX_3 "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, numbusy, numcongestion, numnochan);
 			}
 			*to = 0;
 			return NULL;
@@ -436,7 +436,7 @@
 			if (ast_test_flag(o, DIAL_STILLGOING) && o->chan && (o->chan->_state == AST_STATE_UP)) {
 				if (!peer) {
 					if (option_verbose > 2)
-						ast_verbose( VERBOSE_PREFIX_3 "%s answered %s\n", o->chan->name, in->name);
+						ast_verbose(VERBOSE_PREFIX_3 "%s answered %s\n", o->chan->name, in->name);
 					peer = o->chan;
 					ast_copy_flags(peerflags, o,
 						       OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
@@ -567,7 +567,7 @@
 							break;
 						case AST_CONTROL_BUSY:
 							if (option_verbose > 2)
-								ast_verbose( VERBOSE_PREFIX_3 "%s is busy\n", o->chan->name);
+								ast_verbose(VERBOSE_PREFIX_3 "%s is busy\n", o->chan->name);
 							in->hangupcause = o->chan->hangupcause;
 							ast_hangup(o->chan);
 							o->chan = NULL;
@@ -576,7 +576,7 @@
 							break;
 						case AST_CONTROL_CONGESTION:
 							if (option_verbose > 2)
-								ast_verbose( VERBOSE_PREFIX_3 "%s is circuit-busy\n", o->chan->name);
+								ast_verbose(VERBOSE_PREFIX_3 "%s is circuit-busy\n", o->chan->name);
 							in->hangupcause = o->chan->hangupcause;
 							ast_hangup(o->chan);
 							o->chan = NULL;
@@ -585,7 +585,7 @@
 							break;
 						case AST_CONTROL_RINGING:
 							if (option_verbose > 2)
-								ast_verbose( VERBOSE_PREFIX_3 "%s is ringing\n", o->chan->name);
+								ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", o->chan->name);
 							if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
 								ast_indicate(in, AST_CONTROL_RINGING);
 								(*sentringing)++;
@@ -593,18 +593,18 @@
 							break;
 						case AST_CONTROL_PROGRESS:
 							if (option_verbose > 2)
-								ast_verbose ( VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", o->chan->name,in->name);
+								ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", o->chan->name,in->name);
 							if (!ast_test_flag(outgoing, OPT_RINGBACK))
 								ast_indicate(in, AST_CONTROL_PROGRESS);
 							break;
 						case AST_CONTROL_VIDUPDATE:
 							if (option_verbose > 2)
-								ast_verbose ( VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", o->chan->name,in->name);
+								ast_verbose (VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", o->chan->name,in->name);
 							ast_indicate(in, AST_CONTROL_VIDUPDATE);
 							break;
 						case AST_CONTROL_PROCEEDING:
 							if (option_verbose > 2)
-								ast_verbose ( VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", o->chan->name,in->name);
+								ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", o->chan->name,in->name);
 							if (!ast_test_flag(outgoing, OPT_RINGBACK))
 								ast_indicate(in, AST_CONTROL_PROCEEDING);
 							break;
@@ -625,28 +625,30 @@
 						case -1:
 							if (!ast_test_flag(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
 								if (option_verbose > 2)
-									ast_verbose( VERBOSE_PREFIX_3 "%s stopped sounds\n", o->chan->name);
+									ast_verbose(VERBOSE_PREFIX_3 "%s stopped sounds\n", o->chan->name);
 								ast_indicate(in, -1);
 								(*sentringing) = 0;
 							}
 							break;
 						default:
-							ast_log(LOG_DEBUG, "Dunno what to do with control type %d\n", f->subclass);
+							if (option_debug)
+								ast_log(LOG_DEBUG, "Dunno what to do with control type %d\n", f->subclass);
 						}
 					} else if (single && (f->frametype == AST_FRAME_VOICE) && 
 								!(ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK))) {
 						if (ast_write(in, f)) 
-							ast_log(LOG_DEBUG, "Unable to forward frame\n");
+							ast_log(LOG_WARNING, "Unable to forward voice frame\n");
 					} else if (single && (f->frametype == AST_FRAME_IMAGE) && 
 								!(ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK))) {
 						if (ast_write(in, f))
-							ast_log(LOG_DEBUG, "Unable to forward image\n");
+							ast_log(LOG_WARNING, "Unable to forward image\n");
 					} else if (single && (f->frametype == AST_FRAME_TEXT) && 
 								!(ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK))) {
 						if (ast_write(in, f))
-							ast_log(LOG_DEBUG, "Unable to text\n");
+							ast_log(LOG_WARNING, "Unable to send text\n");
 					} else if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag(outgoing, DIAL_NOFORWARDHTML))
-						ast_channel_sendhtml(in, f->subclass, f->data, f->datalen);
+						if(ast_channel_sendhtml(in, f->subclass, f->data, f->datalen) == -1)
+							ast_log(LOG_WARNING, "Unable to send URL\n");
 
 					ast_frfree(f);
 				} else {
@@ -668,7 +670,7 @@
 #endif
 			if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
 				/* Got hung up */
-				*to=-1;
+				*to = -1;
 				strcpy(status, "CANCEL");
 				if (f)
 					ast_frfree(f);
@@ -679,7 +681,7 @@
 				if (ast_test_flag(peerflags, OPT_DTMF_EXIT)) {
 					context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
 					if (onedigit_goto(in, context, (char) f->subclass, 1)) {
-						if (option_verbose > 3)
+						if (option_verbose > 2)
 							ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
 						*to=0;
 						*result = f->subclass;
@@ -690,8 +692,8 @@
 				}
 
 				if (ast_test_flag(peerflags, OPT_CALLER_HANGUP) && 
-						  (f->subclass == '*')) { /* hmm it it not guarenteed to be '*' anymore. */
-					if (option_verbose > 3)
+						  (f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
+					if (option_verbose > 2)
 						ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
 					*to=0;
 					strcpy(status, "CANCEL");
@@ -702,7 +704,8 @@
 
 			/* Forward HTML stuff */
 			if (single && f && (f->frametype == AST_FRAME_HTML) && !ast_test_flag(outgoing, DIAL_NOFORWARDHTML)) 
-				ast_channel_sendhtml(outgoing->chan, f->subclass, f->data, f->datalen);
+				if(ast_channel_sendhtml(outgoing->chan, f->subclass, f->data, f->datalen) == -1)
+					ast_log(LOG_WARNING, "Unable to send URL\n");
 			
 
 			if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF)))  {
@@ -711,13 +714,13 @@
 			}
 			if (single && (f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_VIDUPDATE)) {
 				if (option_verbose > 2)
-					ast_verbose ( VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", in->name,outgoing->chan->name);
+					ast_verbose(VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", in->name,outgoing->chan->name);
 				ast_indicate(outgoing->chan, AST_CONTROL_VIDUPDATE);
 			}
 			ast_frfree(f);
 		}
 		if (!*to && (option_verbose > 2))
-			ast_verbose( VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", orig);
+			ast_verbose(VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", orig);
 	}
 
 	return peer;
@@ -806,7 +809,7 @@
 	if (ast_test_flag(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
 		calldurationlimit = atoi(opt_args[OPT_ARG_DURATION_STOP]);
 		if (option_verbose > 2)
-			ast_verbose(VERBOSE_PREFIX_3 "Setting call duration limit to %d seconds.\n",calldurationlimit);			
+			ast_verbose(VERBOSE_PREFIX_3 "Setting call duration limit to %d seconds.\n", calldurationlimit);			
 	}
 
 	if (ast_test_flag(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
@@ -883,13 +886,13 @@
 			ast_shrink_phone_number(l);
 			if( ast_test_flag(&opts, OPT_PRIVACY) ) {
 				if (option_verbose > 2)
-					ast_verbose( VERBOSE_PREFIX_3  "Privacy DB is '%s', clid is '%s'\n",
+					ast_verbose(VERBOSE_PREFIX_3  "Privacy DB is '%s', clid is '%s'\n",
 						     opt_args[OPT_ARG_PRIVACY], l);
 				privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
 			}
 			else {
 				if (option_verbose > 2)
-					ast_verbose( VERBOSE_PREFIX_3  "Privacy Screening, clid is '%s'\n", l);
+					ast_verbose(VERBOSE_PREFIX_3  "Privacy Screening, clid is '%s'\n", l);
 				privdb_val = AST_PRIVACY_UNKNOWN;
 			}
 		} else {
@@ -902,7 +905,7 @@
 					*tn2 = '=';  /* any other chars to be afraid of? */
 			}
 			if (option_verbose > 2)
-				ast_verbose( VERBOSE_PREFIX_3  "Privacy-- callerid is empty\n");
+				ast_verbose(VERBOSE_PREFIX_3  "Privacy-- callerid is empty\n");
 
 			snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
 			l = callerid;
@@ -916,35 +919,36 @@
 				ast_verbose( VERBOSE_PREFIX_3  "CallerID set (%s); N option set; Screening should be off\n", privcid);
 			privdb_val = AST_PRIVACY_ALLOW;
 		}
-		else if( ast_test_flag(&opts, OPT_SCREEN_NOCLID) && strncmp(privcid,"NOCALLERID",10) == 0 ) {
+		else if(ast_test_flag(&opts, OPT_SCREEN_NOCLID) && strncmp(privcid,"NOCALLERID",10) == 0 ) {
 			if (option_verbose > 2)
 				ast_verbose( VERBOSE_PREFIX_3  "CallerID blank; N option set; Screening should happen; dbval is %d\n", privdb_val);
 		}
 		
-		if( privdb_val == AST_PRIVACY_DENY ) {
-			ast_verbose( VERBOSE_PREFIX_3  "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
+		if(privdb_val == AST_PRIVACY_DENY ) {
+			if (option_verbose > 2)
+				ast_verbose( VERBOSE_PREFIX_3  "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
 			res=0;
 			goto out;
 		}
-		else if( privdb_val == AST_PRIVACY_KILL ) {
+		else if(privdb_val == AST_PRIVACY_KILL ) {
 			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 201);
 			res = 0;
 			goto out; /* Is this right? */
 		}
-		else if( privdb_val == AST_PRIVACY_TORTURE ) {
+		else if(privdb_val == AST_PRIVACY_TORTURE ) {
 			ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 301);
 			res = 0;
 			goto out; /* is this right??? */
 
 		}
-		else if( privdb_val == AST_PRIVACY_UNKNOWN ) {
+		else if(privdb_val == AST_PRIVACY_UNKNOWN ) {
 			/* Get the user's intro, store it in priv-callerintros/$CID, 
 			   unless it is already there-- this should be done before the 
 			   call is actually dialed  */
 
 			/* make sure the priv-callerintros dir exists? */
 
-			snprintf(privintro,sizeof(privintro),"priv-callerintros/%s", privcid);
+			snprintf(privintro,sizeof(privintro), "priv-callerintros/%s", privcid);
 			if( ast_fileexists(privintro,NULL,NULL ) > 0 && strncmp(privcid,"NOCALLERID",10) != 0) {
 				/* the DELUX version of this code would allow this caller the
 				   option to hear and retape their previously recorded intro.
@@ -960,8 +964,8 @@
 
 				*/
 				ast_play_and_record(chan, "priv-recordintro", privintro, 4, "gsm", &duration, 128, 2000, 0);  /* NOTE: I've reduced the total time to 4 sec */
-															/* don't think we'll need a lock removed, we took care of
-															   conflicts by naming the privintro file */
+										/* don't think we'll need a lock removed, we took care of
+										   conflicts by naming the privintro file */
 			}
 		}
 	}
@@ -1011,7 +1015,7 @@
 		tmp->chan = ast_request(tech, chan->nativeformats, numsubst, &cause);
 		if (!tmp->chan) {
 			/* If we can't, just go on to the next call */
-			ast_log(LOG_NOTICE, "Unable to create channel of type '%s' (cause %d - %s)\n", tech, cause, ast_cause2str(cause));
+			ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n", tech, cause, ast_cause2str(cause));
 			HANDLE_CAUSE(cause, chan);
 			cur = rest;
 			if (!cur)
@@ -1202,7 +1206,8 @@
 			number = numsubst;
 		pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
  		if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
- 			ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", args.url);
+			if (option_debug)
+ 				ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", args.url);
  			ast_channel_sendurl( peer, args.url );
  		}
 		if (ast_test_flag(&opts, OPT_PRIVACY) || ast_test_flag(&opts, OPT_SCREENING)) {
@@ -1250,7 +1255,8 @@
 							if( ast_test_flag(&opts, OPT_SCREENING) )
 								res2 = ast_play_and_wait(peer,"screen-callee-options");
 						}
-						/* priv-callee-options script:
+						/*! \page DialPrivacy Dial Privacy scripts
+						\par priv-callee-options script:
 							"Dial 1 if you wish this caller to reach you directly in the future,
 								and immediately connect to their incoming call
 							 Dial 2 if you wish to send this caller to voicemail now and 
@@ -1258,18 +1264,16 @@
 							 Dial 3 to send this callerr to the torture menus, now and forevermore.
 							 Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
 							 Dial 5 to allow this caller to come straight thru to you in the future,
-						but right now, just this once, send them to voicemail."
-						*/
-				
-						/* screen-callee-options script:
+								but right now, just this once, send them to voicemail."
+						\par screen-callee-options script:
 							"Dial 1 if you wish to immediately connect to the incoming call
 							 Dial 2 if you wish to send this caller to voicemail.
 							 Dial 3 to send this callerr to the torture menus.
 							 Dial 4 to send this caller to a simple "go away" menu.
 						*/
-						if( !res2 || res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2 > '5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4') ) {
+						if(!res2 || res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2 > '5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4') ) {
 							/* invalid option */
-							res2 = ast_play_and_wait(peer,"vm-sorry");
+							res2 = ast_play_and_wait(peer, "vm-sorry");
 						}
 						loopcount++; /* give the callee a couple chances to make a choice */
 					} while( (!res2 || res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2 > '5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4')) && loopcount < 2 );
@@ -1279,7 +1283,7 @@
 				case '1':
 					if( ast_test_flag(&opts, OPT_PRIVACY) ) {
 						if (option_verbose > 2)
-							ast_verbose( VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to ALLOW\n",
+							ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to ALLOW\n",
 								     opt_args[OPT_ARG_PRIVACY], privcid);
 						ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_ALLOW);
 					}
@@ -1287,7 +1291,7 @@
 				case '2':
 					if( ast_test_flag(&opts, OPT_PRIVACY) ) {
 						if (option_verbose > 2)
-							ast_verbose( VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to DENY\n",
+							ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to DENY\n",
 								     opt_args[OPT_ARG_PRIVACY], privcid);
 						ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_DENY);
 					}
@@ -1304,7 +1308,7 @@
 				case '3':
 					if( ast_test_flag(&opts, OPT_PRIVACY) ) {
 						if (option_verbose > 2)
-							ast_verbose( VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to TORTURE\n",
+							ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to TORTURE\n",
 								     opt_args[OPT_ARG_PRIVACY], privcid);
 						ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_TORTURE);
 					}
@@ -1323,7 +1327,7 @@
 				case '4':
 					if( ast_test_flag(&opts, OPT_PRIVACY) ) {
 						if (option_verbose > 2)
-							ast_verbose( VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to KILL\n",
+							ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to KILL\n",
 								     opt_args[OPT_ARG_PRIVACY], privcid);
 						ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_KILL);
 					}
@@ -1342,7 +1346,7 @@
 				case '5':
 					if( ast_test_flag(&opts, OPT_PRIVACY) ) {
 						if (option_verbose > 2)
-							ast_verbose( VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to ALLOW\n",
+							ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to ALLOW\n",
 								     opt_args[OPT_ARG_PRIVACY], privcid);
 						ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_ALLOW);
 						if (ast_test_flag(&opts, OPT_MUSICBACK)) {
@@ -1361,8 +1365,7 @@
 					/* well, there seems basically two choices. Just patch the caller thru immediately,
 				                  or,... put 'em thru to voicemail. */
 					/* since the callee may have hung up, let's do the voicemail thing, no database decision */
-					if (option_verbose > 2)
-						ast_log(LOG_NOTICE,"privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
+					ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
 					if (ast_test_flag(&opts, OPT_MUSICBACK)) {
 						ast_moh_stop(chan);
 					} else if (ast_test_flag(&opts, OPT_RINGBACK)) {
@@ -1385,10 +1388,10 @@
 				   just clog things up, and it's not useful information, not being tied to a CID */
 				if( strncmp(privcid,"NOCALLERID",10) == 0 || ast_test_flag(&opts, OPT_SCREEN_NOINTRO) ) {
 					ast_filedelete(privintro, NULL);
-					if( ast_fileexists(privintro,NULL,NULL ) > 0 )
-						ast_log(LOG_NOTICE,"privacy: ast_filedelete didn't do its job on %s\n", privintro);
+					if( ast_fileexists(privintro, NULL, NULL ) > 0 )
+						ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", privintro);
 					else if (option_verbose > 2)
-						ast_verbose( VERBOSE_PREFIX_3 "Successfully deleted %s intro file\n", privintro);
+						ast_verbose(VERBOSE_PREFIX_3 "Successfully deleted %s intro file\n", privintro);
 				}
 			}
 		}
@@ -1483,12 +1486,12 @@
 					} else if (!strcasecmp(macro_result, "ABORT")) {
 						/* Hangup both ends unless the caller has the g flag */
 						res = -1;
-					} else if (!strncasecmp(macro_result, "GOTO:",5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
+					} else if (!strncasecmp(macro_result, "GOTO:", 5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
 						res = -1;
 						/* perform a transfer to a new extension */
-						if (strchr(macro_transfer_dest,'^')) { /* context^exten^priority*/
+						if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
 							/* no brainer mode... substitute ^ with | and feed it to builtin goto */
-							for (res=0;res<strlen(macro_transfer_dest);res++)
+							for (res = 0; res < strlen(macro_transfer_dest); res++)
 								if (macro_transfer_dest[res] == '^')
 									macro_transfer_dest[res] = '|';
 
@@ -1510,12 +1513,12 @@
 			}
 			if (!ast_strlen_zero(dtmfcalled)) { 
 				if (option_verbose > 2)
-					ast_verbose(VERBOSE_PREFIX_3 "Sending DTMF '%s' to the called party.\n",dtmfcalled);
+					ast_verbose(VERBOSE_PREFIX_3 "Sending DTMF '%s' to the called party.\n", dtmfcalled);
 				res = ast_dtmf_stream(peer,chan,dtmfcalled,250);
 			}
 			if (!ast_strlen_zero(dtmfcalling)) {
 				if (option_verbose > 2)
-					ast_verbose(VERBOSE_PREFIX_3 "Sending DTMF '%s' to the calling party.\n",dtmfcalling);
+					ast_verbose(VERBOSE_PREFIX_3 "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
 				res = ast_dtmf_stream(chan,peer,dtmfcalling,250);
 			}
 		}
@@ -1590,10 +1593,11 @@
 	}
 	hanguptree(outgoing, NULL);
 	pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
-	ast_log(LOG_DEBUG, "Exiting with DIALSTATUS=%s.\n", status);
+	if (option_debug)
+		ast_log(LOG_DEBUG, "Exiting with DIALSTATUS=%s.\n", status);
 	
 	if ((ast_test_flag(peerflags, OPT_GO_ON)) && (!chan->_softhangup) && (res != AST_PBX_KEEPALIVE))
-		res=0;
+		res = 0;
 	
 	LOCAL_USER_REMOVE(u);    
 	

Modified: team/oej/astum/asterisk.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/asterisk.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/asterisk.c (original)
+++ team/oej/astum/asterisk.c Tue Jan 17 13:14:24 2006
@@ -130,7 +130,7 @@
 
 /*! \brief Welcome message when starting a CLI interface */
 #define WELCOME_MESSAGE \
-	ast_verbose("Asterisk " ASTERISK_VERSION ", Copyright (C) 1999 - 2005 Digium, Inc. and others.\n"); \
+	ast_verbose("Asterisk " ASTERISK_VERSION ", Copyright (C) 1999 - 2006 Digium, Inc. and others.\n"); \
 	ast_verbose("Created by Mark Spencer <markster at digium.com>\n"); \
 	ast_verbose("Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.\n"); \
 	ast_verbose("This is free software, with components licensed under the GNU General Public\n"); \
@@ -2129,6 +2129,10 @@
 		}
 	}
 
+	if (ast_opt_console && !option_verbose) 
+		ast_verbose("[ Reading Master Configuration ]");
+	ast_readconfig();
+
 	if (ast_opt_dump_core) {
 		struct rlimit l;
 		memset(&l, 0, sizeof(l));
@@ -2138,10 +2142,6 @@
 			ast_log(LOG_WARNING, "Unable to disable core size resource limit: %s\n", strerror(errno));
 		}
 	}
-
-	if (ast_opt_console && !option_verbose) 
-		ast_verbose("[ Reading Master Configuration ]");
-	ast_readconfig();
 
 	if ((!rungroup) && !ast_strlen_zero(ast_config_AST_RUN_GROUP))
 		rungroup = ast_config_AST_RUN_GROUP;

Modified: team/oej/astum/channel.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/channel.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/channel.c (original)
+++ team/oej/astum/channel.c Tue Jan 17 13:14:24 2006
@@ -3460,8 +3460,6 @@
 		    !nativefailed && !c0->monitor && !c1->monitor &&
 		    !c0->spies && !c1->spies) {
 			/* Looks like they share a bridge method and nothing else is in the way */
-			if (option_verbose > 2) 
-				ast_verbose(VERBOSE_PREFIX_3 "Attempting native bridge of %s and %s\n", c0->name, c1->name);
 			ast_set_flag(c0, AST_FLAG_NBRIDGE);
 			ast_set_flag(c1, AST_FLAG_NBRIDGE);
 			if ((res = c0->tech->bridge(c0, c1, config->flags, fo, rc, to)) == AST_BRIDGE_COMPLETE) {
@@ -3493,7 +3491,9 @@
 			case AST_BRIDGE_RETRY:
 				continue;
 			default:
-				ast_log(LOG_WARNING, "Private bridge between %s and %s failed\n", c0->name, c1->name);
+				if (option_verbose > 2)
+					ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s ended\n",
+						    c0->name, c1->name);
 				/* fallthrough */
 			case AST_BRIDGE_FAILED_NOWARN:
 				nativefailed++;

Modified: team/oej/astum/channels/chan_local.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/channels/chan_local.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/channels/chan_local.c (original)
+++ team/oej/astum/channels/chan_local.c Tue Jan 17 13:14:24 2006
@@ -390,7 +390,7 @@
 	struct local_pvt *cur, *prev=NULL;
 	struct ast_channel *ochan = NULL;
 	int glaredetect;
-	char *status;
+	const char *status;
 
 	ast_mutex_lock(&p->lock);
 	isoutbound = IS_OUTBOUND(ast, p);

Modified: team/oej/astum/channels/chan_misdn.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/channels/chan_misdn.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/channels/chan_misdn.c (original)
+++ team/oej/astum/channels/chan_misdn.c Tue Jan 17 13:14:24 2006
@@ -2023,6 +2023,9 @@
 		misdn_lib_bridge(ch1->bc,ch2->bc);
 	}
 	
+	if (option_verbose > 2) 
+		ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
+
 	chan_misdn_log(1, ch1->bc->port, "* Makeing Native Bridge between %s and %s\n", ch1->bc->oad, ch2->bc->oad);
   
 	while(1) {

Modified: team/oej/astum/channels/chan_vpb.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/channels/chan_vpb.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/channels/chan_vpb.c (original)
+++ team/oej/astum/channels/chan_vpb.c Tue Jan 17 13:14:24 2006
@@ -475,6 +475,9 @@
 		if (option_verbose>1) 
 			ast_verbose(VERBOSE_PREFIX_2 "%s: vpb_bridge: Bridging call entered with [%s, %s]\n",p0->dev, c0->name, c1->name);
 	}
+
+	if (option_verbose > 2) 
+		ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
 
 	#ifdef HALF_DUPLEX_BRIDGE
 

Modified: team/oej/astum/channels/chan_zap.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/channels/chan_zap.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/channels/chan_zap.c (original)
+++ team/oej/astum/channels/chan_zap.c Tue Jan 17 13:14:24 2006
@@ -3207,6 +3207,9 @@
 		return AST_BRIDGE_FAILED;
 	}
 	
+	if (option_verbose > 2) 
+		ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
+
 	if (!(flags & AST_BRIDGE_DTMF_CHANNEL_0) && (oi0 == SUB_REAL))
 		disable_dtmf_detect(op0);
 

Modified: team/oej/astum/cli.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/cli.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/cli.c (original)
+++ team/oej/astum/cli.c Tue Jan 17 13:14:24 2006
@@ -951,6 +951,19 @@
 
 static int handle_help(int fd, int argc, char *argv[]);
 
+static char * complete_help(char *text, char *word, int pos, int state)
+{
+	/* skip first 4 or 5 chars, "help "*/
+	int l = strlen(text);
+
+	if (l > 5)
+		l = 5;
+	text += l;
+
+	/* XXX watch out, should stop to the non-generator parts */
+	return __ast_cli_generator(text, word, state, 0); /* Don't lock as we are already locked */
+}
+
 static struct ast_cli_entry builtins[] = {
 	/* Keep alphabetized, with longer matches first (example: abcd before abc) */
 	{ { "_command", "complete", NULL }, handle_commandcomplete, "Command complete", commandcomplete_help },
@@ -959,7 +972,7 @@
 	{ { "debug", "channel", NULL }, handle_debugchan, "Enable debugging on a channel", debugchan_help, complete_ch_3 },
 	{ { "debug", "level", NULL }, handle_debuglevel, "Set global debug level", debuglevel_help },
 	{ { "group", "show", "channels", NULL }, group_show_channels, "Show active channels with group(s)", group_show_channels_help},
-	{ { "help", NULL }, handle_help, "Display help list, or specific help on a command", help_help },
+	{ { "help", NULL }, handle_help, "Display help list, or specific help on a command", help_help, complete_help },
 	{ { "load", NULL }, handle_load, "Load a dynamic module by name", load_help, complete_fn },
 	{ { "no", "debug", "channel", NULL }, handle_nodebugchan, "Disable debugging on a channel", nodebugchan_help, complete_ch_4 },
 	{ { "reload", NULL }, handle_reload, "Reload configuration", reload_help, complete_mod_2 },

Modified: team/oej/astum/include/asterisk/doxyref.h
URL: http://svn.digium.com/view/asterisk/team/oej/astum/include/asterisk/doxyref.h?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/include/asterisk/doxyref.h (original)
+++ team/oej/astum/include/asterisk/doxyref.h Tue Jan 17 13:14:24 2006
@@ -289,7 +289,7 @@
 /*! \page Config_enum ENUM Configuration
  * \section enumconf enum.conf
  * \arg See also \ref enumreadme
- * \arg Implemented in \ref app_enumlookup.c and \ref enum.c
+ * \arg Implemented in \ref func_enum.c and \ref enum.c
  * \verbinclude enum.conf.sample
  */
 

Modified: team/oej/astum/logger.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/logger.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/logger.c (original)
+++ team/oej/astum/logger.c Tue Jan 17 13:14:24 2006
@@ -465,9 +465,9 @@
 	ast_mutex_unlock(&loglock);
 
 	filesize_reload_needed = 0;
-
+	
+	init_logger_chain();
 	queue_log_init();
-	init_logger_chain();
 
 	if (logfiles.event_log) {
 		if (eventlog) {
@@ -594,11 +594,11 @@
 	ast_cli_register(&rotate_logger_cli);
 	ast_cli_register(&logger_show_channels_cli);
 
+	/* create log channels */
+	init_logger_chain();
+
 	/* initialize queue logger */
 	queue_log_init();
-
-	/* create log channels */
-	init_logger_chain();
 
 	/* create the eventlog */
 	if (logfiles.event_log) {

Modified: team/oej/astum/manager.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/manager.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/manager.c (original)
+++ team/oej/astum/manager.c Tue Jan 17 13:14:24 2006
@@ -1533,7 +1533,7 @@
 			if (timestampevents) {
 				now = ast_tvnow();
 				ast_build_string(&tmp_next, &tmp_left, "Timestamp: %ld.%06lu\r\n",
-						 now.tv_sec, now.tv_usec);
+						 now.tv_sec, (unsigned long) now.tv_usec);
 			}
 			va_start(ap, fmt);
 			ast_build_string_va(&tmp_next, &tmp_left, fmt, ap);

Modified: team/oej/astum/res/res_features.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/res/res_features.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/res/res_features.c (original)
+++ team/oej/astum/res/res_features.c Tue Jan 17 13:14:24 2006
@@ -1272,7 +1272,7 @@
 			src = chan;
 		else if ((monitor_exec = pbx_builtin_getvar_helper(peer, "AUTO_MONITOR")))
 			src = peer;
-		if (src) {
+		if (monitor_app && src) {
 			char *tmp = ast_strdupa(monitor_exec);
 			if (tmp) {
 				pbx_exec(src, monitor_app, tmp, 1);

Modified: team/oej/astum/rtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/astum/rtp.c?rev=8139&r1=8138&r2=8139&view=diff
==============================================================================
--- team/oej/astum/rtp.c (original)
+++ team/oej/astum/rtp.c Tue Jan 17 13:14:24 2006
@@ -1655,6 +1655,9 @@
 		}
 	}
 
+	if (option_verbose > 2) 
+		ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
+
 	/* Ok, we should be able to redirect the media. Start with one channel */
 	if (pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE))) 
 		ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);



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