[asterisk-commits] oej: branch oej/adb-appleraision-1.8-mark-2 r414994 - in /team/oej/adb-appler...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jun 2 09:20:46 CDT 2014
Author: oej
Date: Mon Jun 2 09:20:36 2014
New Revision: 414994
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=414994
Log:
Hey ho
Modified:
team/oej/adb-appleraision-1.8-mark-2/ (props changed)
team/oej/adb-appleraision-1.8-mark-2/UPGRADE.txt
team/oej/adb-appleraision-1.8-mark-2/addons/chan_ooh323.c
team/oej/adb-appleraision-1.8-mark-2/addons/ooh323cDriver.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_adsiprog.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_chanspy.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_dial.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_dumpchan.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_festival.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_forkcdr.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_getcpeid.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_jack.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_minivm.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_mixmonitor.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_playback.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_queue.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_readexten.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_saycounted.c (props changed)
team/oej/adb-appleraision-1.8-mark-2/apps/app_speech_utils.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_stack.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_transfer.c
team/oej/adb-appleraision-1.8-mark-2/apps/app_voicemail.c
team/oej/adb-appleraision-1.8-mark-2/cel/cel_custom.c
team/oej/adb-appleraision-1.8-mark-2/cel/cel_manager.c
team/oej/adb-appleraision-1.8-mark-2/cel/cel_odbc.c
team/oej/adb-appleraision-1.8-mark-2/cel/cel_pgsql.c
team/oej/adb-appleraision-1.8-mark-2/cel/cel_radius.c
team/oej/adb-appleraision-1.8-mark-2/channels/chan_multicast_rtp.c (contents, props changed)
team/oej/adb-appleraision-1.8-mark-2/channels/chan_sip.c
team/oej/adb-appleraision-1.8-mark-2/channels/sig_ss7.c (props changed)
team/oej/adb-appleraision-1.8-mark-2/channels/sip/config_parser.c
team/oej/adb-appleraision-1.8-mark-2/channels/sip/dialplan_functions.c
team/oej/adb-appleraision-1.8-mark-2/channels/sip/include/sip.h
team/oej/adb-appleraision-1.8-mark-2/channels/sip/reqresp_parser.c
team/oej/adb-appleraision-1.8-mark-2/pbx/dundi-parser.c
team/oej/adb-appleraision-1.8-mark-2/pbx/pbx_config.c
team/oej/adb-appleraision-1.8-mark-2/pbx/pbx_dundi.c
team/oej/adb-appleraision-1.8-mark-2/pbx/pbx_loopback.c
team/oej/adb-appleraision-1.8-mark-2/pbx/pbx_lua.c
team/oej/adb-appleraision-1.8-mark-2/pbx/pbx_spool.c
team/oej/adb-appleraision-1.8-mark-2/tests/test_dlinklists.c
team/oej/adb-appleraision-1.8-mark-2/tests/test_expr.c (props changed)
team/oej/adb-appleraision-1.8-mark-2/tests/test_func_file.c (props changed)
team/oej/adb-appleraision-1.8-mark-2/tests/test_linkedlists.c
team/oej/adb-appleraision-1.8-mark-2/tests/test_locale.c (props changed)
team/oej/adb-appleraision-1.8-mark-2/tests/test_poll.c (props changed)
team/oej/adb-appleraision-1.8-mark-2/tests/test_substitution.c
Propchange: team/oej/adb-appleraision-1.8-mark-2/
------------------------------------------------------------------------------
--- automerge-email (original)
+++ automerge-email Mon Jun 2 09:20:36 2014
@@ -1,1 +1,0 @@
-oej at edvina.net
Propchange: team/oej/adb-appleraision-1.8-mark-2/
('svnmerge-integrated' removed)
Modified: team/oej/adb-appleraision-1.8-mark-2/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/UPGRADE.txt?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/UPGRADE.txt (original)
+++ team/oej/adb-appleraision-1.8-mark-2/UPGRADE.txt Mon Jun 2 09:20:36 2014
@@ -18,7 +18,72 @@
===
===========================================================
-<<<<<<< .working
+from 1.8.27.0 to 1.8.28.0:
+* The asterisk command line -I option and the asterisk.conf internal_timing
+ option are removed and always enabled if any timing module is loaded.
+* SIP (chan_sip) accounts dialed through a Local channel will now properly
+ hide the "1 missed call" if one of the other dialed accounts picks up the
+ call.
+
+* Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
+ objects will emit additional debug information to the refs log file located
+ in the standard Asterisk log file directory. This log file is useful in
+ tracking down object leaks and other reference counting issues. Prior to
+ this version, this option was only available by modifying the source code
+ directly. This change also includes a new script, refcounter.py, in the
+ contrib folder that will process the refs log file.
+
+from 1.8.26.0 to 1.8.27.0:
+* res_fax now returns the correct rates for V.27ter (4800 or 9600 bit/s).
+ Because of this the default settings would not load, so the minrate (minimum
+ transmission rate) option was changed to default to 4800 since that is the
+ minimum rate for v.27 which is included in the default modem options.
+
+* When communicating with a peer on an Asterisk 1.4 or earlier system, the
+ chan_iax2 parameter 'connectedline' must be set to "no" in iax.conf. This
+ prevents an incompatible connected line frame from an Astersik 1.8 or later
+ system from causing a hangup in an Asterisk 1.4 or earlier system. Note that
+ this particular incompatibility has always existed between 1.4 and 1.8 and
+ later versions; this upgrade note is simply informing users of its existance.
+
+* A compatibility setting, allow_empty_string_in_nontext, has been added to
+ res_odbc.conf. When enabled (default behavior), empty column values are
+ stored as empty strings during realtime updates. Disabling this option
+ causes empty column values to be stored as NULLs for non-text columns.
+
+ Disable it for PostgreSQL backends in order to avoid errors caused by
+ updating integer columns with an empty string instead of NULL
+ (sipppeers,sipregs)
+
+from 1.8.23.0 to 1.8.24.0:
+* res_agi will now properly indicate if there was an error in streaming an
+ audio file. The result code will be -1 and the result returned from the
+ the function will be RESULT_FAILURE instead of the prior behavior of always
+ returning RESULT_SUCCESS even if there was an error.
+
+* The option "register_retry_403" has been added to chan_sip to work around
+ servers that are known to erroneously send 403 in response to valid
+ REGISTER requests and allows Asterisk to continue attepmting to connect.
+ Due to a failed merge, this option is present, but non-functional until 1.8.26.0.
+
+* Certain dialplan functions have been marked as 'dangerous', and may only be
+ executed from the dialplan. Execution from extenal sources (AMI's GetVar and
+ SetVar actions; etc.) may be inhibited by setting live_dangerously in the
+ [options] section of asterisk.conf to no. SHELL(), channel locking, and direct
+ file read/write functions are marked as dangerous. DB_DELETE() and
+ REALTIME_DESTROY() are marked as dangerous for reads, but can now safely
+ accept writes (which ignore the provided value).
+
+from 1.8.22.0 to 1.8.23.0:
+* The default settings for chan_sip are now overriden properly by the general
+ settings in sip.conf. Please look over your settings upon upgrading.
+
+* It is now possible to play the Queue prompts to the first user waiting in a call queue.
+ Note that this may impact the ability for agents to talk with users, as a prompt may
+ still be playing when an agent connects to the user. This ability is disabled by
+ default but can be enabled on an individual queue using the 'announce-to-first-user'
+ option.
+
from 1.8.21.0 to 1.8.22.0:
* Added the 'n' option to MeetMe to prevent application of the DENOISE function
to a channel joining a conference. Some channel drivers that vary the number
@@ -59,114 +124,6 @@
the makefile in favor of using native optimization suppport when available.
BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".
-=======
-from 1.8.27.0 to 1.8.28.0:
-* The asterisk command line -I option and the asterisk.conf internal_timing
- option are removed and always enabled if any timing module is loaded.
-* SIP (chan_sip) accounts dialed through a Local channel will now properly
- hide the "1 missed call" if one of the other dialed accounts picks up the
- call.
-
-* Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
- objects will emit additional debug information to the refs log file located
- in the standard Asterisk log file directory. This log file is useful in
- tracking down object leaks and other reference counting issues. Prior to
- this version, this option was only available by modifying the source code
- directly. This change also includes a new script, refcounter.py, in the
- contrib folder that will process the refs log file.
-
-from 1.8.26.0 to 1.8.27.0:
-* res_fax now returns the correct rates for V.27ter (4800 or 9600 bit/s).
- Because of this the default settings would not load, so the minrate (minimum
- transmission rate) option was changed to default to 4800 since that is the
- minimum rate for v.27 which is included in the default modem options.
-
-* When communicating with a peer on an Asterisk 1.4 or earlier system, the
- chan_iax2 parameter 'connectedline' must be set to "no" in iax.conf. This
- prevents an incompatible connected line frame from an Astersik 1.8 or later
- system from causing a hangup in an Asterisk 1.4 or earlier system. Note that
- this particular incompatibility has always existed between 1.4 and 1.8 and
- later versions; this upgrade note is simply informing users of its existance.
-
-* A compatibility setting, allow_empty_string_in_nontext, has been added to
- res_odbc.conf. When enabled (default behavior), empty column values are
- stored as empty strings during realtime updates. Disabling this option
- causes empty column values to be stored as NULLs for non-text columns.
-
- Disable it for PostgreSQL backends in order to avoid errors caused by
- updating integer columns with an empty string instead of NULL
- (sipppeers,sipregs)
-
-from 1.8.23.0 to 1.8.24.0:
-* res_agi will now properly indicate if there was an error in streaming an
- audio file. The result code will be -1 and the result returned from the
- the function will be RESULT_FAILURE instead of the prior behavior of always
- returning RESULT_SUCCESS even if there was an error.
-
-* The option "register_retry_403" has been added to chan_sip to work around
- servers that are known to erroneously send 403 in response to valid
- REGISTER requests and allows Asterisk to continue attepmting to connect.
- Due to a failed merge, this option is present, but non-functional until 1.8.26.0.
-
-* Certain dialplan functions have been marked as 'dangerous', and may only be
- executed from the dialplan. Execution from extenal sources (AMI's GetVar and
- SetVar actions; etc.) may be inhibited by setting live_dangerously in the
- [options] section of asterisk.conf to no. SHELL(), channel locking, and direct
- file read/write functions are marked as dangerous. DB_DELETE() and
- REALTIME_DESTROY() are marked as dangerous for reads, but can now safely
- accept writes (which ignore the provided value).
-
-from 1.8.22.0 to 1.8.23.0:
-* The default settings for chan_sip are now overriden properly by the general
- settings in sip.conf. Please look over your settings upon upgrading.
-
-* It is now possible to play the Queue prompts to the first user waiting in a call queue.
- Note that this may impact the ability for agents to talk with users, as a prompt may
- still be playing when an agent connects to the user. This ability is disabled by
- default but can be enabled on an individual queue using the 'announce-to-first-user'
- option.
-
-from 1.8.21.0 to 1.8.22.0:
-* Added the 'n' option to MeetMe to prevent application of the DENOISE function
- to a channel joining a conference. Some channel drivers that vary the number
- of audio samples in a voice frame will experience significant quality problems
- if a denoiser is attached to the channel; this option gives them the ability
- to remove the denoiser without having to unload func_speex.
-
-* The Registry AMI event for SIP registrations will now always include the
- Username field. A previous bug fix missed an instance where it was not
- included; that has been corrected in this release.
-
-from 1.8.20.0 to 1.8.20.1:
-* Asterisk would previously not output certain error messages when a remote
- console attempted to connect to Asterisk and no instance of Asterisk was
- running. This error message is displayed on stderr; as a result, some
- initialization scripts that used remote consoles to test for the presence
- of a running Asterisk instance started to display erroneous error messages.
- The init.d scripts and the safe_asterisk have been updated in the contrib
- folder to account for this.
-
-from 1.8.19 to 1.8.20:
-* Asterisk has always had code to ignore dash '-' characters that are not
- part of a character set in the dialplan extensions. The code now
- consistently ignores these characters when matching dialplan extensions.
-
-from 1.8.18 to 1.8.19:
-* Queue strategy rrmemory now has a predictable order similar to strategy
- rrordered. Members will be called in the order that they are added to the
- queue.
-
-From 1.8.13 to 1.8.14:
-* permitdirectmedia/denydirectmedia now controls whether peers can be
- bridged via directmedia by comparing the ACL to the bridging peer's
- address rather than its own address.
-
-From 1.8.12 to 1.8.13:
-* The complex processor detection and optimization has been removed from
- the makefile in favor of using native optimization suppport when available.
- BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".
-
->>>>>>> .merge-right.r414880
From 1.8.11 to 1.8.12:
* In AEL dialplans, the "h" extension will now be inherited from prior
calling contexts, just as it had in 1.4. If you have created an AEL
Modified: team/oej/adb-appleraision-1.8-mark-2/addons/chan_ooh323.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/addons/chan_ooh323.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/addons/chan_ooh323.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/addons/chan_ooh323.c Mon Jun 2 09:20:36 2014
@@ -23,6 +23,7 @@
***/
#include "chan_ooh323.h"
+#include "asterisk/paths.h"
#include <math.h>
#define FORMAT_STRING_SIZE 512
@@ -30,7 +31,7 @@
/* Defaults */
#define DEFAULT_CONTEXT "default"
#define DEFAULT_H323ID "Asterisk PBX"
-#define DEFAULT_LOGFILE "/var/log/asterisk/h323_log"
+#define DEFAULT_LOGFILE "h323_log"
#define DEFAULT_H323ACCNT "ast_h323"
/* Flags */
@@ -286,6 +287,8 @@
int onCallCleared(ooCallData *call);
void onModeChanged(ooCallData *call, int t38mode);
+extern OOH323EndPoint gH323ep;
+
static char gLogFile[256] = DEFAULT_LOGFILE;
static int gPort = 1720;
static char gIP[20];
@@ -629,6 +632,7 @@
ooh323_destroy(p);
ast_mutex_unlock(&iflock);
ast_log(LOG_ERROR, "Destination format is not supported\n");
+ *cause = AST_CAUSE_INVALID_NUMBER_FORMAT;
return NULL;
}
@@ -671,6 +675,10 @@
ast_mutex_unlock(&p->lock);
ooh323_destroy(p);
ast_mutex_unlock(&iflock);
+ return NULL;
+ } else if (gH323ep.gkClient && gH323ep.gkClient->state != GkClientRegistered) {
+ ast_log(LOG_ERROR, "Gatekeeper client is configured but not registered\n");
+ *cause = AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
return NULL;
}
p->g729onlyA = g729onlyA;
@@ -2562,7 +2570,7 @@
}
/* Inintialize everything to default */
- strcpy(gLogFile, DEFAULT_LOGFILE);
+ snprintf(gLogFile, sizeof(gLogFile), "%s/%s", ast_config_AST_LOG_DIR, DEFAULT_LOGFILE);
gPort = 1720;
gIP[0] = '\0';
strcpy(gCallerID, DEFAULT_H323ID);
Modified: team/oej/adb-appleraision-1.8-mark-2/addons/ooh323cDriver.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/addons/ooh323cDriver.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/addons/ooh323cDriver.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/addons/ooh323cDriver.c Mon Jun 2 09:20:36 2014
@@ -100,8 +100,10 @@
pfds[0].fd = mycthread->thePipe[0];
pfds[0].events = POLLIN;
ooSocketPoll(pfds, 1, SEC_TO_HOLD_THREAD * 1000);
- if (ooPDRead(pfds, 1, mycthread->thePipe[0]))
+ if (ooPDRead(pfds, 1, mycthread->thePipe[0])) {
res = read(mycthread->thePipe[0], &c, 1);
+ (void) res;/* Shut up compiler: Set but not used and unused return value of read. */
+ }
ast_mutex_lock(&callThreadsLock);
ast_mutex_lock(&mycthread->lock);
@@ -182,6 +184,7 @@
cur->inUse = TRUE;
cur->call = call;
res = write(cur->thePipe[1], &c, 1);
+ (void) res;/* Shut up compiler: Set but not used and unused return value of write. */
ast_mutex_unlock(&cur->lock);
}
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_adsiprog.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_adsiprog.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_adsiprog.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_adsiprog.c Mon Jun 2 09:20:36 2014
@@ -202,7 +202,7 @@
if (!(argtype & ARG_NUMBER))
return -1;
/* Octal value */
- if (sscanf(src, "%30o", (int *)out) != 1)
+ if (sscanf(src, "%30o", (unsigned *)out) != 1)
return -1;
if (argtype & ARG_STRING) {
/* Convert */
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_chanspy.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_chanspy.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_chanspy.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_chanspy.c Mon Jun 2 09:20:36 2014
@@ -158,7 +158,9 @@
</option>
<option name="x">
<argument name="digit" required="true">
- <para>Specify a DTMF digit that can be used to exit the application.</para>
+ <para>Specify a DTMF digit that can be used to exit the application while actively
+ spying on a channel. If there is no channel being spied on, the DTMF digit will be
+ ignored.</para>
</argument>
</option>
<option name="X">
@@ -295,7 +297,9 @@
</option>
<option name="x">
<argument name="digit" required="true">
- <para>Specify a DTMF digit that can be used to exit the application.</para>
+ <para>Specify a DTMF digit that can be used to exit the application while actively
+ spying on a channel. If there is no channel being spied on, the DTMF digit will be
+ ignored.</para>
</argument>
</option>
<option name="X">
@@ -363,7 +367,7 @@
OPTION_NAME = (1 << 12), /* Say the name of the person on whom we will spy */
OPTION_DTMF_SWITCH_MODES = (1 << 13), /* Allow numeric DTMF to switch between chanspy modes */
OPTION_DTMF_EXIT = (1 << 14), /* Set DTMF to exit, added for DAHDIScan integration */
- OPTION_DTMF_CYCLE = (1 << 15), /* Custom DTMF for cycling next avaliable channel, (default is '*') */
+ OPTION_DTMF_CYCLE = (1 << 15), /* Custom DTMF for cycling next available channel, (default is '*') */
OPTION_DAHDI_SCAN = (1 << 16), /* Scan groups in DAHDIScan mode */
OPTION_STOP = (1 << 17),
OPTION_EXITONHANGUP = (1 << 18), /* Hang up when the spied-on channel hangs up. */
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_dial.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_dial.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_dial.c Mon Jun 2 09:20:36 2014
@@ -1470,7 +1470,7 @@
/* Fall through */
case AST_FRAME_TEXT:
if (single && ast_write(in, f)) {
- ast_log(LOG_WARNING, "Unable to write frametype: %d\n",
+ ast_log(LOG_WARNING, "Unable to write frametype: %u\n",
f->frametype);
}
break;
@@ -1575,7 +1575,7 @@
case AST_FRAME_DTMF_BEGIN:
case AST_FRAME_DTMF_END:
if (ast_write(o->chan, f)) {
- ast_log(LOG_WARNING, "Unable to forward frametype: %d\n",
+ ast_log(LOG_WARNING, "Unable to forward frametype: %u\n",
f->frametype);
}
break;
@@ -2760,8 +2760,9 @@
/* perform a transfer to a new extension */
if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
replace_macro_delimiter(macro_transfer_dest);
- if (!ast_parseable_goto(chan, macro_transfer_dest))
- ast_set_flag64(peerflags, OPT_GO_ON);
+ }
+ if (!ast_parseable_goto(chan, macro_transfer_dest)) {
+ ast_set_flag64(peerflags, OPT_GO_ON);
}
}
}
@@ -2872,8 +2873,9 @@
/* perform a transfer to a new extension */
if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
replace_macro_delimiter(gosub_transfer_dest);
- if (!ast_parseable_goto(chan, gosub_transfer_dest))
- ast_set_flag64(peerflags, OPT_GO_ON);
+ }
+ if (!ast_parseable_goto(chan, gosub_transfer_dest)) {
+ ast_set_flag64(peerflags, OPT_GO_ON);
}
}
}
@@ -3023,7 +3025,9 @@
}
ast_channel_early_bridge(chan, NULL);
- hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
+ /* When dialing local channels, the hangupcause of the parent channel
+ * tells us whether the call was answered elsewhere. */
+ hanguptree(outgoing, NULL, chan->hangupcause == AST_CAUSE_ANSWERED_ELSEWHERE ? 1 : 0);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
senddialendevent(chan, pa.status);
ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_dumpchan.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_dumpchan.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_dumpchan.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_dumpchan.c Mon Jun 2 09:20:36 2014
@@ -49,7 +49,7 @@
</synopsis>
<syntax>
<parameter name="level">
- <para>Minimun verbose level</para>
+ <para>Minimum verbose level</para>
</parameter>
</syntax>
<description>
@@ -107,7 +107,7 @@
"RDNIS= %s\n"
"Parkinglot= %s\n"
"Language= %s\n"
- "State= %s (%d)\n"
+ "State= %s (%u)\n"
"Rings= %d\n"
"NativeFormat= %s\n"
"WriteFormat= %s\n"
@@ -117,8 +117,8 @@
"WriteTranscode= %s %s\n"
"ReadTranscode= %s %s\n"
"1stFileDescriptor= %d\n"
- "Framesin= %d %s\n"
- "Framesout= %d %s\n"
+ "Framesin= %u %s\n"
+ "Framesout= %u %s\n"
"TimetoHangup= %ld\n"
"ElapsedTime= %dh%dm%ds\n"
"DirectBridge= %s\n"
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_festival.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_festival.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_festival.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_festival.c Mon Jun 2 09:20:36 2014
@@ -414,7 +414,7 @@
/* Convert to HEX and look if there is any matching file in the cache
directory */
for (i = 0; i < 16; i++) {
- snprintf(koko, sizeof(koko), "%X", MD5Res[i]);
+ snprintf(koko, sizeof(koko), "%X", (unsigned)MD5Res[i]);
strncat(MD5Hex, koko, sizeof(MD5Hex) - strlen(MD5Hex) - 1);
}
readcache = 0;
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_forkcdr.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_forkcdr.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_forkcdr.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_forkcdr.c Mon Jun 2 09:20:36 2014
@@ -239,13 +239,14 @@
{
int res = 0;
char *argcopy = NULL;
+ struct ast_cdr *cdr;
struct ast_flags flags = {0};
char *opts[OPT_ARG_ARRAY_SIZE];
AST_DECLARE_APP_ARGS(arglist,
AST_APP_ARG(options);
);
- if (!chan->cdr) {
+ if (!(cdr = chan->cdr)) {
ast_log(LOG_WARNING, "Channel does not have a CDR\n");
return 0;
}
@@ -261,7 +262,10 @@
if (!ast_strlen_zero(data)) {
int keepvars = ast_test_flag(&flags, OPT_KEEPVARS) ? 1 : 0;
- ast_set2_flag(chan->cdr, keepvars, AST_CDR_FLAG_KEEP_VARS);
+ while (cdr->next) {
+ cdr = cdr->next;
+ }
+ ast_set2_flag(cdr, keepvars, AST_CDR_FLAG_KEEP_VARS);
}
ast_cdr_fork(chan, flags, opts[OPT_ARG_VARSET]);
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_getcpeid.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_getcpeid.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_getcpeid.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_getcpeid.c Mon Jun 2 09:20:36 2014
@@ -87,7 +87,9 @@
res = ast_adsi_get_cpeid(chan, cpeid, 0);
if (res > 0) {
gotcpeid = 1;
- ast_verb(3, "Got CPEID of '%02x:%02x:%02x:%02x' on '%s'\n", cpeid[0], cpeid[1], cpeid[2], cpeid[3], chan->name);
+ ast_verb(3, "Got CPEID of '%02x:%02x:%02x:%02x' on '%s'\n",
+ (unsigned)cpeid[0], (unsigned)cpeid[1], (unsigned)cpeid[2],
+ (unsigned)cpeid[3], chan->name);
}
if (res > -1) {
strcpy(data[1], "Measuring CPE...");
@@ -101,7 +103,9 @@
}
if (res > -1) {
if (gotcpeid)
- snprintf(data[1], 80, "CPEID: %02x:%02x:%02x:%02x", cpeid[0], cpeid[1], cpeid[2], cpeid[3]);
+ snprintf(data[1], 80, "CPEID: %02x:%02x:%02x:%02x",
+ (unsigned)cpeid[0], (unsigned)cpeid[1],
+ (unsigned)cpeid[2], (unsigned)cpeid[3]);
else
strcpy(data[1], "CPEID Unknown");
if (gotgeometry)
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_jack.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_jack.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_jack.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_jack.c Mon Jun 2 09:20:36 2014
@@ -951,6 +951,11 @@
{
int res;
+ if (!chan) {
+ ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
+ return -1;
+ }
+
if (!strcasecmp(value, "on"))
res = enable_jack_hook(chan, data);
else if (!strcasecmp(value, "off"))
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_minivm.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_minivm.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_minivm.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_minivm.c Mon Jun 2 09:20:36 2014
@@ -1390,7 +1390,7 @@
}
}
- fprintf(p, "Message-ID: <Asterisk-%d-%s-%d-%s>\n", (unsigned int)ast_random(), vmu->username, (int)getpid(), who);
+ fprintf(p, "Message-ID: <Asterisk-%u-%s-%d-%s>\n", (unsigned int)ast_random(), vmu->username, (int)getpid(), who);
if (ast_strlen_zero(vmu->email)) {
snprintf(email, sizeof(email), "%s@%s", vmu->username, vmu->domain);
@@ -1441,7 +1441,7 @@
fprintf(p, "MIME-Version: 1.0\n");
/* Something unique. */
- snprintf(bound, sizeof(bound), "voicemail_%s%d%d", vmu->username, (int)getpid(), (unsigned int)ast_random());
+ snprintf(bound, sizeof(bound), "voicemail_%s%d%u", vmu->username, (int)getpid(), (unsigned int)ast_random());
fprintf(p, "Content-Type: multipart/mixed; boundary=\"%s\"\n\n\n", bound);
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_mixmonitor.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_mixmonitor.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_mixmonitor.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_mixmonitor.c Mon Jun 2 09:20:36 2014
@@ -381,6 +381,8 @@
ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
mixmonitor_free(mixmonitor);
+
+ ast_module_unref(ast_module_info->self);
return NULL;
}
@@ -414,7 +416,7 @@
return 0;
}
-static void launch_monitor_thread(struct ast_channel *chan, const char *filename, unsigned int flags,
+static int launch_monitor_thread(struct ast_channel *chan, const char *filename, unsigned int flags,
int readvol, int writevol, const char *post_process)
{
pthread_t thread;
@@ -442,26 +444,26 @@
/* Pre-allocate mixmonitor structure and spy */
if (!(mixmonitor = ast_calloc(1, len))) {
- return;
+ return -1;
}
/* Setup the actual spy before creating our thread */
if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type)) {
mixmonitor_free(mixmonitor);
- return;
+ return -1;
}
/* Copy over flags and channel name */
mixmonitor->flags = flags;
if (!(mixmonitor->autochan = ast_autochan_setup(chan))) {
mixmonitor_free(mixmonitor);
- return;
+ return -1;
}
if (setup_mixmonitor_ds(mixmonitor, chan)) {
ast_autochan_destroy(mixmonitor->autochan);
mixmonitor_free(mixmonitor);
- return;
+ return -1;
}
mixmonitor->name = (char *) mixmonitor + sizeof(*mixmonitor);
strcpy(mixmonitor->name, chan->name);
@@ -485,10 +487,10 @@
mixmonitor_spy_type, chan->name);
ast_audiohook_destroy(&mixmonitor->audiohook);
mixmonitor_free(mixmonitor);
- return;
- }
-
- ast_pthread_create_detached_background(&thread, NULL, mixmonitor_thread, mixmonitor);
+ return -1;
+ }
+
+ return ast_pthread_create_detached_background(&thread, NULL, mixmonitor_thread, mixmonitor);
}
static int mixmonitor_exec(struct ast_channel *chan, const char *data)
@@ -567,7 +569,12 @@
ast_mkdir(tmp, 0777);
pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
- launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process);
+
+ /* If launch_monitor_thread works, the module reference must not be released until it is finished. */
+ ast_module_ref(ast_module_info->self);
+ if (launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process)) {
+ ast_module_unref(ast_module_info->self);
+ }
return 0;
}
@@ -583,7 +590,7 @@
ast_mutex_lock(&mixmonitor_ds->lock);
- /* closing the filestream here guarantees the file is avaliable to the dialplan
+ /* closing the filestream here guarantees the file is available to the dialplan
* after calling StopMixMonitor */
mixmonitor_ds_close_fs(mixmonitor_ds);
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_playback.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_playback.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_playback.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_playback.c Mon Jun 2 09:20:36 2014
@@ -211,6 +211,10 @@
s = x + 1;
ast_debug(2, "value is <%s>\n", s);
n = ast_var_assign("SAY", s);
+ if (!n) {
+ ast_log(LOG_ERROR, "Memory allocation error in do_say\n");
+ return -1;
+ }
AST_LIST_INSERT_HEAD(&head, n, entries);
/* scan the body, one piece at a time */
Modified: team/oej/adb-appleraision-1.8-mark-2/apps/app_queue.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/apps/app_queue.c?view=diff&rev=414994&r1=414993&r2=414994
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/apps/app_queue.c (original)
+++ team/oej/adb-appleraision-1.8-mark-2/apps/app_queue.c Mon Jun 2 09:20:36 2014
@@ -205,10 +205,10 @@
connected to a queue member.</para>
</parameter>
<parameter name="macro">
- <para>Will run a macro on the calling party's channel once they are connected to a queue member.</para>
+ <para>Will run a macro on the called party's channel (the queue member) once the parties are connected.</para>
</parameter>
<parameter name="gosub">
- <para>Will run a gosub on the calling party's channel once they are connected to a queue member.</para>
+ <para>Will run a gosub on the called party's channel (the queue member) once the parties are connected.</para>
</parameter>
<parameter name="rule">
<para>Will cause the queue's defaultrule to be overridden by the rule specified.</para>
@@ -685,6 +685,7 @@
<syntax>
</syntax>
<description>
+ <para>Show queues information.</para>
</description>
</manager>
<manager name="QueueStatus" language="en_US">
@@ -693,10 +694,15 @@
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Queue" />
- <parameter name="Member" />
+ <parameter name="Queue">
+ <para>Limit the response to the status of the specified queue.</para>
+ </parameter>
+ <parameter name="Member">
+ <para>Limit the response to the status of the specified member.</para>
+ </parameter>
</syntax>
<description>
+ <para>Check the status of one or more queues.</para>
</description>
</manager>
<manager name="QueueSummary" language="en_US">
@@ -705,9 +711,12 @@
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Queue" />
+ <parameter name="Queue">
+ <para>Queue for which the summary is requested.</para>
+ </parameter>
</syntax>
<description>
+ <para>Request the manager to send a QueueSummary event.</para>
</description>
</manager>
<manager name="QueueAdd" language="en_US">
@@ -716,11 +725,21 @@
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Queue" required="true" />
- <parameter name="Interface" required="true" />
- <parameter name="Penalty" />
- <parameter name="Paused" />
- <parameter name="MemberName" />
+ <parameter name="Queue" required="true">
+ <para>Queue's name.</para>
+ </parameter>
+ <parameter name="Interface" required="true">
+ <para>The name of the interface (tech/name) to add to the queue.</para>
+ </parameter>
+ <parameter name="Penalty">
+ <para>A penalty (number) to apply to this member. Asterisk will distribute calls to members with higher penalties only after attempting to distribute calls to those with lower penalty.</para>
+ </parameter>
+ <parameter name="Paused">
+ <para>To pause or not the member initially (true/false or 1/0).</para>
+ </parameter>
+ <parameter name="MemberName">
+ <para>Text alias for the interface.</para>
+ </parameter>
<parameter name="StateInterface" />
</syntax>
<description>
@@ -732,8 +751,12 @@
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Queue" required="true" />
- <parameter name="Interface" required="true" />
+ <parameter name="Queue" required="true">
+ <para>The name of the queue to take action on.</para>
+ </parameter>
+ <parameter name="Interface" required="true">
+ <para>The interface (tech/name) to remove from queue.</para>
+ </parameter>
</syntax>
<description>
</description>
@@ -744,12 +767,21 @@
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Interface" required="true" />
- <parameter name="Paused" required="true" />
- <parameter name="Queue" />
- <parameter name="Reason" />
+ <parameter name="Interface" required="true">
+ <para>The name of the interface (tech/name) to pause or unpause.</para>
+ </parameter>
+ <parameter name="Paused" required="true">
+ <para>Pause or unpause the interface. Set to 'true' to pause the member or 'false' to unpause.</para>
+ </parameter>
+ <parameter name="Queue">
+ <para>The name of the queue in which to pause or unpause this member. If not specified, the member will be paused or unpaused in all the queues it is a member of.</para>
+ </parameter>
+ <parameter name="Reason">
+ <para>Text description, returned in the event QueueMemberPaused.</para>
+ </parameter>
</syntax>
<description>
+ <para>Pause or unpause a member in a queue.</para>
</description>
</manager>
<manager name="QueueLog" language="en_US">
@@ -773,11 +805,18 @@
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Interface" required="true" />
- <parameter name="Penalty" required="true" />
- <parameter name="Queue" />
+ <parameter name="Interface" required="true">
+ <para>The interface (tech/name) of the member whose penalty to change.</para>
+ </parameter>
+ <parameter name="Penalty" required="true">
+ <para>The new penalty (number) for the member. Must be nonnegative.</para>
+ </parameter>
+ <parameter name="Queue">
+ <para>If specified, only set the penalty for the member of this queue. Otherwise, set the penalty for the member in all queues to which the member belongs.</para>
+ </parameter>
</syntax>
<description>
+ <para>Change the penalty of a queue member</para>
</description>
</manager>
<manager name="QueueRule" language="en_US">
@@ -786,9 +825,12 @@
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Rule" />
+ <parameter name="Rule">
+ <para>The name of the rule in queuerules.conf whose contents to list.</para>
+ </parameter>
</syntax>
<description>
+ <para>List queue rules defined in queuerules.conf</para>
</description>
</manager>
<manager name="QueueReload" language="en_US">
@@ -797,20 +839,25 @@
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
- <parameter name="Queue" />
+ <parameter name="Queue">
+ <para>The name of the queue to take action on. If no queue name is specified, then all queues are affected.</para>
+ </parameter>
<parameter name="Members">
+ <para>Whether to reload the queue's members.</para>
<enumlist>
<enum name="yes" />
<enum name="no" />
</enumlist>
</parameter>
<parameter name="Rules">
+ <para>Whether to reload queuerules.conf</para>
<enumlist>
[... 1499 lines stripped ...]
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