[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
........
................
r8127 | mogorman | 2006-01-17 18:25:53 +0100 (Tue, 17 Jan 2006) | 2 lines
Added tab completion for help. bug 6074
................
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)
................
r8135 | bweschke | 2006-01-17 19:31:03 +0100 (Tue, 17 Jan 2006) | 3 lines
Fix compiler warning.
................
r8136 | oej | 2006-01-17 19:54:56 +0100 (Tue, 17 Jan 2006) | 3 lines
- Logging clean up
- Whitespace removed and added, formatting fixed
................
r8137 | oej | 2006-01-17 19:56:57 +0100 (Tue, 17 Jan 2006) | 2 lines
Extra comma causing compilation errors...
................
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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