[svn-commits] tilghman: branch 1.6.0 r160389 - in /branches/1.6.0: ./ apps/ channels/ inclu...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Dec 2 16:56:37 CST 2008
Author: tilghman
Date: Tue Dec 2 16:56:36 2008
New Revision: 160389
URL: http://svn.digium.com/view/asterisk?view=rev&rev=160389
Log:
Merged revisions 152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r152969 | tilghman | 2008-10-30 15:35:46 -0500 (Thu, 30 Oct 2008) | 10 lines
Merged revisions 152958 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) | 3 lines
Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
(Closes issue #13400)
........
................
r153122 | tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10 lines
Merged revisions 153114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) | 3 lines
Turn off qualify on uncached realtime peers.
(Closes issue #13383)
........
................
r154264 | tilghman | 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines
Recorded merge of revisions 154263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines
Make the monitor thread non-detached, so it can be joined (suggested by Russell
on -dev list).
........
................
r154268 | rmudgett | 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) | 11 lines
Merged revisions 154266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) | 4 lines
JIRA ABE-1703
mISDN sets the channel to the wrong state when it receives
the indication AST_CONTROL_RINGING.
........
................
r154366 | tilghman | 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) | 16 lines
Merged revisions 154365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines
On busy systems, it's possible for the values checked within a single line
of code to change, unless the structure is locked to ensure a consistent
state.
(closes issue #13717)
Reported by: kowalma
Patches:
20081102__bug13717.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
........
................
r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) | 14 lines
Merged revisions 155398 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines
Clarify error message.
(closes issue #13809)
Reported by: denke
Patches:
20081104__bug13809.diff.txt uploaded by Corydon76 (license 14)
Tested by: denke
........
................
r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) | 22 lines
Merged revisions 155861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines
Channel drivers assume that when their indicate callback
is invoked, that the channel on which the callback was called
is locked. This patch corrects an instance in chan_agent where
a channel's indicate callback is called directly without first
locking the channel.
This was leading to some observed locking issues in chan_local,
but considering that all channel drivers operate under the
same expectations, the generic fix in chan_agent is the right
way to go.
AST-126
........
................
r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) | 15 lines
Merged revisions 156164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines
Move the sanity check that makes sure "always fork" is not set along with the
console option to be after the code that reads options from asterisk.conf.
This resolves a situation where Asterisk can start taking up 100% when
misconfigured.
(Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to
figure out what was causing the 100% CPU problem.)
........
................
r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) | 13 lines
Merged revisions 156294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines
If the SLA thread is not started, then reload causes a memory leak.
(closes issue #13889)
Reported by: eliel
Patches:
app_meetme.c.patch uploaded by eliel (license 64)
........
................
r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) | 14 lines
Merged revisions 156688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines
Provide more space for all the data which can appear in an originating
channel name.
(closes issue #13398)
Reported by: bamby
Patches:
manager.c.diff uploaded by bamby (license 430)
........
................
r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines
Merged revisions 156755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines
ast_waitfordigit() requires that the channel be up, for no good logical
reason. This prevents While/EndWhile from working within the "h"
extension.
Reported by: jgalarneau (for ABE C.2)
Fixed by: me
........
................
r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines
Merged revisions 158053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
........
................
r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines
Merged revisions 158071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines
We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
........
................
r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines
Merged revisions 158539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines
When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock
........
................
r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines
Merged revisions 158600 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines
The passed extension may not be the same in the list as the current entry,
because we strip spaces when copying the extension into the structure.
Therefore, use the copied item to place the item into the list.
(found by lmadsen on -dev, fixed by me)
........
................
r159276 | tilghman | 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines
Merged revisions 159269 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines
Don't try to send a response on a NULL pvt.
(closes issue #13919)
Reported by: barthpbx
Patches:
chan_iax2.c.patch uploaded by eliel (license 64)
Tested by: barthpbx
........
................
Modified:
branches/1.6.0/ (props changed)
branches/1.6.0/apps/app_dial.c
branches/1.6.0/apps/app_meetme.c
branches/1.6.0/apps/app_while.c
branches/1.6.0/channels/chan_agent.c
branches/1.6.0/channels/chan_h323.c
branches/1.6.0/channels/chan_iax2.c
branches/1.6.0/channels/chan_misdn.c
branches/1.6.0/channels/chan_sip.c
branches/1.6.0/channels/chan_skinny.c
branches/1.6.0/include/asterisk/astobj2.h
branches/1.6.0/main/asterisk.c
branches/1.6.0/main/astobj2.c
branches/1.6.0/main/manager.c
branches/1.6.0/main/pbx.c
Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/apps/app_dial.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/apps/app_dial.c (original)
+++ branches/1.6.0/apps/app_dial.c Tue Dec 2 16:56:36 2008
@@ -1583,6 +1583,9 @@
/* Again, keep going even if there's an error */
ast_debug(1, "ast call on peer returned %d\n", res);
ast_verb(3, "Couldn't call %s\n", numsubst);
+ if (tc->hangupcause) {
+ chan->hangupcause = tc->hangupcause;
+ }
ast_hangup(tc);
tc = NULL;
ast_free(tmp);
Modified: branches/1.6.0/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/apps/app_meetme.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/apps/app_meetme.c (original)
+++ branches/1.6.0/apps/app_meetme.c Tue Dec 2 16:56:36 2008
@@ -1378,6 +1378,10 @@
struct sla_trunk_ref *trunk_ref, struct sla_station *station, int lock)
{
struct sla_event *event;
+
+ if (sla.thread == AST_PTHREADT_NULL) {
+ return;
+ }
if (!(event = ast_calloc(1, sizeof(*event))))
return;
Modified: branches/1.6.0/apps/app_while.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/apps/app_while.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/apps/app_while.c (original)
+++ branches/1.6.0/apps/app_while.c Tue Dec 2 16:56:36 2008
@@ -160,11 +160,16 @@
return -1;
}
+#if 0
/* dont want run away loops if the chan isn't even up
this is up for debate since it slows things down a tad ......
+
+ Debate is over... this prevents While/EndWhile from working
+ within the "h" extension. Not good.
*/
if (ast_waitfordigit(chan,1) < 0)
return -1;
+#endif
for (x=0;;x++) {
if (get_index(chan, prefix, x)) {
Modified: branches/1.6.0/channels/chan_agent.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_agent.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/channels/chan_agent.c (original)
+++ branches/1.6.0/channels/chan_agent.c Tue Dec 2 16:56:36 2008
@@ -657,9 +657,15 @@
struct agent_pvt *p = ast->tech_pvt;
int res = -1;
ast_mutex_lock(&p->lock);
- if (p->chan && !ast_check_hangup(p->chan))
- res = p->chan->tech->indicate ? p->chan->tech->indicate(p->chan, condition, data, datalen) : -1;
- else
+ if (p->chan && !ast_check_hangup(p->chan)) {
+ while (ast_channel_trylock(p->chan)) {
+ ast_channel_unlock(ast);
+ usleep(1);
+ ast_channel_lock(ast);
+ }
+ res = p->chan->tech->indicate ? p->chan->tech->indicate(p->chan, condition, data, datalen) : -1;
+ ast_channel_unlock(p->chan);
+ } else
res = 0;
ast_mutex_unlock(&p->lock);
return res;
Modified: branches/1.6.0/channels/chan_h323.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_h323.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/channels/chan_h323.c (original)
+++ branches/1.6.0/channels/chan_h323.c Tue Dec 2 16:56:36 2008
@@ -2609,7 +2609,7 @@
pthread_kill(monitor_thread, SIGURG);
} else {
/* Start a new monitor */
- if (ast_pthread_create_detached_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {
+ if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {
monitor_thread = AST_PTHREADT_NULL;
ast_mutex_unlock(&monlock);
ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
@@ -3301,9 +3301,9 @@
}
if (!ast_mutex_lock(&monlock)) {
if ((monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {
- /* this causes a seg, anyone know why? */
- if (monitor_thread != pthread_self())
+ if (monitor_thread != pthread_self()) {
pthread_cancel(monitor_thread);
+ }
pthread_kill(monitor_thread, SIGURG);
pthread_join(monitor_thread, NULL);
}
Modified: branches/1.6.0/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_iax2.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/channels/chan_iax2.c (original)
+++ branches/1.6.0/channels/chan_iax2.c Tue Dec 2 16:56:36 2008
@@ -1664,11 +1664,20 @@
/* This will occur on the first response to a message that we initiated,
* such as a PING. */
+ if (dcallno) {
+ ast_mutex_lock(&iaxsl[dcallno]);
+ }
if (callno && dcallno && iaxs[dcallno] && !iaxs[dcallno]->peercallno && match(sin, callno, dcallno, iaxs[dcallno], check_dcallno)) {
iaxs[dcallno]->peercallno = callno;
res = dcallno;
store_by_peercallno(iaxs[dcallno]);
+ if (!res || !return_locked) {
+ ast_mutex_unlock(&iaxsl[dcallno]);
+ }
return res;
+ }
+ if (dcallno) {
+ ast_mutex_unlock(&iaxsl[dcallno]);
}
#ifdef IAX_OLD_FIND
@@ -6977,7 +6986,7 @@
/* Make sure our call still exists, an INVAL at the right point may make it go away */
if (!iaxs[callno]) {
- res = 0;
+ res = -1;
goto return_unref;
}
Modified: branches/1.6.0/channels/chan_misdn.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_misdn.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/channels/chan_misdn.c (original)
+++ branches/1.6.0/channels/chan_misdn.c Tue Dec 2 16:56:36 2008
@@ -2479,7 +2479,7 @@
}
chan_misdn_log(3, p->bc->port, " --> * SEND: State Ring pid:%d\n", p->bc ? p->bc->pid : -1);
- ast_setstate(ast, AST_STATE_RINGING);
+ ast_setstate(ast, AST_STATE_RING);
if ( !p->bc->nt && (p->originator == ORG_MISDN) && !p->incoming_early_audio )
chan_misdn_log(2, p->bc->port, " --> incoming_early_audio off\n");
Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Tue Dec 2 16:56:36 2008
@@ -4280,8 +4280,11 @@
res = update_call_counter(p, INC_CALL_RINGING);
- if (res == -1)
+ if (res == -1) {
return res;
+ } else {
+ ast->hangupcause = AST_CAUSE_USER_BUSY;
+ }
p->callingpres = ast->cid.cid_pres;
p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
@@ -15266,7 +15269,7 @@
break;
case 200: /* 200 OK */
if (!r) {
- ast_log(LOG_WARNING, "Got 200 OK on REGISTER that isn't a register\n");
+ ast_log(LOG_WARNING, "Got 200 OK on REGISTER, but there isn't a registry entry for '%s' (we probably already got the OK)\n", S_OR(p->peername, p->username));
p->needdestroy = 1;
return 0;
}
@@ -15445,6 +15448,20 @@
Fix assigned to Rizzo :-)
*/
/* check_via_response(p, req); */
+
+ /* RFC 3261 Section 15 specifies that if we receive a 408 or 481
+ * in response to a BYE, then we should end the current dialog
+ * and session. There is no mention in the spec of other 4XX responses,
+ * but it is known that at least one phone manufacturer potentially
+ * will send a 404 in response to a BYE, so we'll be liberal in what
+ * we accept and end the dialog and session if we receive any 4XX
+ * response to a BYE.
+ */
+ if (resp >= 400 && resp < 500 && sipmethod == SIP_BYE) {
+ p->needdestroy = 1;
+ return;
+ }
+
if (p->relatedpeer && p->method == SIP_OPTIONS) {
/* We don't really care what the response is, just that it replied back.
Well, as long as it's not a 100 response... since we might
@@ -20704,6 +20721,14 @@
peer->maxms = default_qualify ? default_qualify : DEFAULT_MAXMS;
} else if (sscanf(v->value, "%d", &peer->maxms) != 1) {
ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno);
+ peer->maxms = 0;
+ }
+ if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && peer->maxms > 0) {
+ /* This would otherwise cause a network storm, where the
+ * qualify response refreshes the peer from the database,
+ * which in turn causes another qualify to be sent, ad
+ * infinitum. */
+ ast_log(LOG_WARNING, "Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer '%s'\n", peer->name);
peer->maxms = 0;
}
} else if (!strcasecmp(v->name, "qualifyfreq")) {
Modified: branches/1.6.0/channels/chan_skinny.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_skinny.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/channels/chan_skinny.c (original)
+++ branches/1.6.0/channels/chan_skinny.c Tue Dec 2 16:56:36 2008
@@ -5747,7 +5747,7 @@
sessions = s;
ast_mutex_unlock(&sessionlock);
- if (ast_pthread_create_detached(&tcp_thread, NULL, skinny_session, s)) {
+ if (ast_pthread_create(&tcp_thread, NULL, skinny_session, s)) {
destroy_session(s);
}
}
Modified: branches/1.6.0/include/asterisk/astobj2.h
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/include/asterisk/astobj2.h?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/include/asterisk/astobj2.h (original)
+++ branches/1.6.0/include/asterisk/astobj2.h Tue Dec 2 16:56:36 2008
@@ -211,6 +211,7 @@
#define ao2_unlock(a) _ao2_unlock(a, __FILE__, __PRETTY_FUNCTION__, __LINE__, #a)
int _ao2_unlock(void *a, const char *file, const char *func, int line, const char *var);
#endif
+int ao2_trylock(void *user_data);
/*!
*
Modified: branches/1.6.0/main/asterisk.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/main/asterisk.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/main/asterisk.c (original)
+++ branches/1.6.0/main/asterisk.c Tue Dec 2 16:56:36 2008
@@ -3028,11 +3028,6 @@
if (ast_opt_console && !option_verbose)
ast_verbose("[ Booting...\n");
- if (ast_opt_always_fork && (ast_opt_remote || ast_opt_console)) {
- ast_log(LOG_WARNING, "'alwaysfork' is not compatible with console or remote console mode; ignored\n");
- ast_clear_flag(&ast_options, AST_OPT_FLAG_ALWAYS_FORK);
- }
-
/* For remote connections, change the name of the remote connection.
* We do this for the benefit of init scripts (which need to know if/when
* the main asterisk process has died yet). */
@@ -3043,8 +3038,10 @@
}
}
- if (ast_opt_console && !option_verbose)
+ if (ast_opt_console && !option_verbose) {
ast_verbose("[ Reading Master Configuration ]\n");
+ }
+
ast_readconfig();
if (ast_opt_remote && remotesock != NULL)
@@ -3052,6 +3049,11 @@
if (!ast_language_is_prefix && !ast_opt_remote)
ast_log(LOG_WARNING, "The 'languageprefix' option in asterisk.conf is deprecated; in a future release it will be removed, and your sound files will need to be organized in the 'new style' language layout.\n");
+
+ if (ast_opt_always_fork && (ast_opt_remote || ast_opt_console)) {
+ ast_log(LOG_WARNING, "'alwaysfork' is not compatible with console or remote console mode; ignored\n");
+ ast_clear_flag(&ast_options, AST_OPT_FLAG_ALWAYS_FORK);
+ }
if (ast_opt_dump_core) {
struct rlimit l;
Modified: branches/1.6.0/main/astobj2.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/main/astobj2.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/main/astobj2.c (original)
+++ branches/1.6.0/main/astobj2.c Tue Dec 2 16:56:36 2008
@@ -168,6 +168,21 @@
#else
return __ast_pthread_mutex_unlock(file, line, func, var, &p->priv_data.lock);
#endif
+}
+
+int ao2_trylock(void *user_data)
+{
+ struct astobj2 *p = INTERNAL_OBJ(user_data);
+ int ret;
+
+ if (p == NULL)
+ return -1;
+ ret = ast_mutex_trylock(&p->priv_data.lock);
+#ifdef AO2_DEBUG
+ if (!ret)
+ ast_atomic_fetchadd_int(&ao2.total_locked, 1);
+#endif
+ return ret;
}
/*
Modified: branches/1.6.0/main/manager.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/main/manager.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/main/manager.c (original)
+++ branches/1.6.0/main/manager.c Tue Dec 2 16:56:36 2008
@@ -2059,7 +2059,8 @@
/*! \brief helper function for originate */
struct fast_originate_helper {
char tech[AST_MAX_EXTENSION];
- char data[AST_MAX_EXTENSION];
+ /*! data can contain a channel name, extension number, username, password, etc. */
+ char data[512];
int timeout;
int format; /*!< Codecs used for a call */
char app[AST_MAX_APP];
Modified: branches/1.6.0/main/pbx.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/main/pbx.c?view=diff&rev=160389&r1=160388&r2=160389
==============================================================================
--- branches/1.6.0/main/pbx.c (original)
+++ branches/1.6.0/main/pbx.c Tue Dec 2 16:56:36 2008
@@ -6795,7 +6795,7 @@
}
res = 0; /* some compilers will think it is uninitialized otherwise */
for (e = con->root; e; el = e, e = e->next) { /* scan the extension list */
- res = ext_cmp(e->exten, extension);
+ res = ext_cmp(e->exten, tmp->exten);
if (res == 0) { /* extension match, now look at cidmatch */
if (!e->matchcid && !tmp->matchcid)
res = 0;
More information about the svn-commits
mailing list