[asterisk-commits] file: branch file/res_xmpp r368711 - in /team/file/res_xmpp: ./ apps/ build_t...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jun 8 13:29:16 CDT 2012
Author: file
Date: Fri Jun 8 13:29:09 2012
New Revision: 368711
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=368711
Log:
Bring up to date.
Added:
team/file/res_xmpp/channels/chan_sip.exports.in
- copied unchanged from r368688, trunk/channels/chan_sip.exports.in
team/file/res_xmpp/configs/app_skel.conf.sample
- copied unchanged from r368688, trunk/configs/app_skel.conf.sample
team/file/res_xmpp/configs/config_test.conf.sample
- copied unchanged from r368688, trunk/configs/config_test.conf.sample
team/file/res_xmpp/funcs/func_presencestate.c
- copied unchanged from r368688, trunk/funcs/func_presencestate.c
team/file/res_xmpp/include/asterisk/app_voicemail.h
- copied unchanged from r368688, trunk/include/asterisk/app_voicemail.h
team/file/res_xmpp/include/asterisk/config_options.h
- copied unchanged from r368688, trunk/include/asterisk/config_options.h
team/file/res_xmpp/include/asterisk/http_websocket.h
- copied unchanged from r368688, trunk/include/asterisk/http_websocket.h
team/file/res_xmpp/include/asterisk/presencestate.h
- copied unchanged from r368688, trunk/include/asterisk/presencestate.h
team/file/res_xmpp/include/asterisk/sip_api.h
- copied unchanged from r368688, trunk/include/asterisk/sip_api.h
team/file/res_xmpp/main/config_options.c
- copied unchanged from r368688, trunk/main/config_options.c
team/file/res_xmpp/main/presencestate.c
- copied unchanged from r368688, trunk/main/presencestate.c
team/file/res_xmpp/res/res_http_websocket.c
- copied unchanged from r368688, trunk/res/res_http_websocket.c
team/file/res_xmpp/res/res_http_websocket.exports.in
- copied unchanged from r368688, trunk/res/res_http_websocket.exports.in
team/file/res_xmpp/tests/test_voicemail_api.c
- copied unchanged from r368688, trunk/tests/test_voicemail_api.c
Modified:
team/file/res_xmpp/ (props changed)
team/file/res_xmpp/CHANGES
team/file/res_xmpp/Makefile
team/file/res_xmpp/UPGRADE.txt
team/file/res_xmpp/apps/app_dial.c
team/file/res_xmpp/apps/app_meetme.c
team/file/res_xmpp/apps/app_mixmonitor.c
team/file/res_xmpp/apps/app_queue.c
team/file/res_xmpp/apps/app_skel.c
team/file/res_xmpp/apps/app_stack.c
team/file/res_xmpp/apps/app_voicemail.c
team/file/res_xmpp/apps/app_voicemail.exports.in
team/file/res_xmpp/build_tools/make_version
team/file/res_xmpp/channels/chan_agent.c
team/file/res_xmpp/channels/chan_dahdi.c
team/file/res_xmpp/channels/chan_iax2.c
team/file/res_xmpp/channels/chan_local.c
team/file/res_xmpp/channels/chan_sip.c
team/file/res_xmpp/channels/chan_skinny.c
team/file/res_xmpp/channels/chan_unistim.c
team/file/res_xmpp/channels/sig_analog.c
team/file/res_xmpp/channels/sig_pri.c
team/file/res_xmpp/channels/sig_ss7.c
team/file/res_xmpp/channels/sip/include/sip.h
team/file/res_xmpp/configs/manager.conf.sample
team/file/res_xmpp/configs/sip.conf.sample
team/file/res_xmpp/configure
team/file/res_xmpp/configure.ac
team/file/res_xmpp/contrib/editors/asterisk.vim
team/file/res_xmpp/contrib/realtime/mysql/voicemail_messages.sql
team/file/res_xmpp/formats/format_ogg_vorbis.c
team/file/res_xmpp/funcs/func_channel.c
team/file/res_xmpp/funcs/func_math.c
team/file/res_xmpp/include/asterisk/app.h
team/file/res_xmpp/include/asterisk/astobj2.h
team/file/res_xmpp/include/asterisk/callerid.h
team/file/res_xmpp/include/asterisk/channel.h
team/file/res_xmpp/include/asterisk/config.h
team/file/res_xmpp/include/asterisk/event_defs.h
team/file/res_xmpp/include/asterisk/file.h
team/file/res_xmpp/include/asterisk/manager.h
team/file/res_xmpp/include/asterisk/message.h
team/file/res_xmpp/include/asterisk/pbx.h
team/file/res_xmpp/include/asterisk/stringfields.h
team/file/res_xmpp/include/asterisk/utils.h
team/file/res_xmpp/main/app.c
team/file/res_xmpp/main/asterisk.c
team/file/res_xmpp/main/asterisk.exports.in
team/file/res_xmpp/main/astobj2.c
team/file/res_xmpp/main/callerid.c
team/file/res_xmpp/main/channel.c
team/file/res_xmpp/main/channel_internal_api.c
team/file/res_xmpp/main/config.c
team/file/res_xmpp/main/event.c
team/file/res_xmpp/main/features.c
team/file/res_xmpp/main/file.c
team/file/res_xmpp/main/manager.c
team/file/res_xmpp/main/message.c
team/file/res_xmpp/main/pbx.c
team/file/res_xmpp/main/tcptls.c
team/file/res_xmpp/main/udptl.c
team/file/res_xmpp/main/utils.c
team/file/res_xmpp/makeopts.in
team/file/res_xmpp/pbx/pbx_config.c
team/file/res_xmpp/res/ael/pval.c
team/file/res_xmpp/res/res_config_odbc.c
team/file/res_xmpp/tests/test_config.c
Propchange: team/file/res_xmpp/
('branch-10-merged' removed)
Propchange: team/file/res_xmpp/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Jun 8 13:29:09 2012
@@ -1,1 +1,1 @@
-/trunk:1-367834
+/trunk:1-368710
Modified: team/file/res_xmpp/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/file/res_xmpp/CHANGES?view=diff&rev=368711&r1=368710&r2=368711
==============================================================================
--- team/file/res_xmpp/CHANGES (original)
+++ team/file/res_xmpp/CHANGES Fri Jun 8 13:29:09 2012
@@ -31,6 +31,13 @@
* The minimum DTMF duration can now be configured in asterisk.conf
as "mindtmfduration". The default value is (as before) set to 80 ms.
(previously it was only available in source code)
+ * Each logging destination and console now have an independent notion of the
+ current verbosity level. Logger.conf now allows an optional argument to
+ the 'verbose' specifier, indicating the level of verbosity sent to that
+ particular logging destination. Additionally, remote consoles now each
+ have their own verbosity level. The command 'core set verbose' will now set
+ a separate level for each remote console without affecting any other
+ console.
CLI Changes
-------------------
@@ -202,16 +209,6 @@
-------------
* Allow the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to be
used within the dynamic weight attribute when specifying a mapping.
-
-Core changes
-------------
- * Each logging destination and console now have an independent notion of the
- current verbosity level. Logger.conf now allows an optional argument to
- the 'verbose' specifier, indicating the level of verbosity sent to that
- particular logging destination. Additionally, remote consoles now each
- have their own verbosity level. The command 'core set verbose' will now set
- a separate level for each remote console without affecting any other
- console.
Dialplan functions
------------------
Modified: team/file/res_xmpp/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/file/res_xmpp/Makefile?view=diff&rev=368711&r1=368710&r2=368711
==============================================================================
--- team/file/res_xmpp/Makefile (original)
+++ team/file/res_xmpp/Makefile Fri Jun 8 13:29:09 2012
@@ -191,6 +191,7 @@
_ASTCFLAGS+=-Wunused
_ASTCFLAGS+=$(AST_DECLARATION_AFTER_STATEMENT)
_ASTCFLAGS+=$(AST_FORTIFY_SOURCE)
+ _ASTCFLAGS+=$(AST_TRAMPOLINES)
_ASTCFLAGS+=-Wundef
_ASTCFLAGS+=-Wmissing-format-attribute
_ASTCFLAGS+=-Wformat=2
Modified: team/file/res_xmpp/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/team/file/res_xmpp/UPGRADE.txt?view=diff&rev=368711&r1=368710&r2=368711
==============================================================================
--- team/file/res_xmpp/UPGRADE.txt (original)
+++ team/file/res_xmpp/UPGRADE.txt Fri Jun 8 13:29:09 2012
@@ -25,6 +25,11 @@
Parking:
- The comebacktoorigin setting must now be set per parking lot. The setting in
the general section will not be applied automatically to each parking lot.
+ - The BLINDTRANSFER channel variable is deleted from a channel when it is
+ bridged to prevent subtle bugs in the parking feature. The channel
+ variable is used by Asterisk internally for the Park application to work
+ properly. If you were using it for your own purposes, copy it to your
+ own channel variable before the channel is bridged.
res_ais:
- Users of res_ais in versions of Asterisk prior to Asterisk 11 must change
Modified: team/file/res_xmpp/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/team/file/res_xmpp/apps/app_dial.c?view=diff&rev=368711&r1=368710&r2=368711
==============================================================================
--- team/file/res_xmpp/apps/app_dial.c (original)
+++ team/file/res_xmpp/apps/app_dial.c Fri Jun 8 13:29:09 2012
@@ -133,8 +133,7 @@
<para>Reset the call detail record (CDR) for this call.</para>
</option>
<option name="c">
- <para>If the Dial() application cancels this call, always set the flag to tell the channel
- driver that the call is answered elsewhere.</para>
+ <para>If the Dial() application cancels this call, always set HANGUPCAUSE to 'answered elsewhere'</para>
</option>
<option name="d">
<para>Allow the calling user to dial a 1 digit extension while waiting for
@@ -727,8 +726,6 @@
/* Hangup any existing lines we have open */
if (outgoing->chan && (outgoing->chan != exception)) {
if (answered_elsewhere) {
- /* The flag is used for local channel inheritance and stuff */
- ast_set_flag(ast_channel_flags(outgoing->chan), AST_FLAG_ANSWERED_ELSEWHERE);
/* This is for the channel drivers */
ast_channel_hangupcause_set(outgoing->chan, AST_CAUSE_ANSWERED_ELSEWHERE);
}
@@ -2515,12 +2512,12 @@
if (outbound_group)
ast_app_group_set_channel(tc, outbound_group);
/* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
- if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_ANSWERED_ELSEWHERE))
- ast_set_flag(ast_channel_flags(tc), AST_FLAG_ANSWERED_ELSEWHERE);
+ if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE)
+ ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
/* Check if we're forced by configuration */
if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
- ast_set_flag(ast_channel_flags(tc), AST_FLAG_ANSWERED_ELSEWHERE);
+ ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
/* Inherit context and extension */
@@ -3079,11 +3076,11 @@
}
ast_channel_early_bridge(chan, NULL);
- hanguptree(&out_chans, NULL, 0); /* In this case, there's no answer anywhere */
+ hanguptree(&out_chans, NULL, ast_channel_hangupcause(chan)==AST_CAUSE_ANSWERED_ELSEWHERE ? 1 : 0 ); /* forward 'answered elsewhere' if we received it */
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
senddialendevent(chan, pa.status);
ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
-
+
if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
if (!ast_tvzero(calldurationlimit))
memset(ast_channel_whentohangup(chan), 0, sizeof(*ast_channel_whentohangup(chan)));
Modified: team/file/res_xmpp/apps/app_meetme.c
URL: http://svnview.digium.com/svn/asterisk/team/file/res_xmpp/apps/app_meetme.c?view=diff&rev=368711&r1=368710&r2=368711
==============================================================================
--- team/file/res_xmpp/apps/app_meetme.c (original)
+++ team/file/res_xmpp/apps/app_meetme.c Fri Jun 8 13:29:09 2012
@@ -2180,12 +2180,12 @@
static void send_talking_event(struct ast_channel *chan, struct ast_conference *conf, struct ast_conf_user *user, int talking)
{
ast_manager_event(chan, EVENT_FLAG_CALL, "MeetmeTalking",
- "Channel: %s\r\n"
- "Uniqueid: %s\r\n"
- "Meetme: %s\r\n"
- "Usernum: %d\r\n"
- "Status: %s\r\n",
- ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no, talking ? "on" : "off");
+ "Channel: %s\r\n"
+ "Uniqueid: %s\r\n"
+ "Meetme: %s\r\n"
+ "Usernum: %d\r\n"
+ "Status: %s\r\n",
+ ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no, talking ? "on" : "off");
}
static void set_user_talking(struct ast_channel *chan, struct ast_conference *conf, struct ast_conf_user *user, int talking, int monitor)
@@ -3144,12 +3144,12 @@
}
ast_manager_event(chan, EVENT_FLAG_CALL, "MeetmeMute",
- "Channel: %s\r\n"
- "Uniqueid: %s\r\n"
- "Meetme: %s\r\n"
- "Usernum: %i\r\n"
- "Status: on\r\n",
- ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no);
+ "Channel: %s\r\n"
+ "Uniqueid: %s\r\n"
+ "Meetme: %s\r\n"
+ "Usernum: %d\r\n"
+ "Status: on\r\n",
+ ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no);
}
/* If I should be un-muted but am not talker, un-mute me */
@@ -3162,12 +3162,12 @@
}
ast_manager_event(chan, EVENT_FLAG_CALL, "MeetmeMute",
- "Channel: %s\r\n"
- "Uniqueid: %s\r\n"
- "Meetme: %s\r\n"
- "Usernum: %i\r\n"
- "Status: off\r\n",
- ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no);
+ "Channel: %s\r\n"
+ "Uniqueid: %s\r\n"
+ "Meetme: %s\r\n"
+ "Usernum: %d\r\n"
+ "Status: off\r\n",
+ ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no);
}
if ((user->adminflags & (ADMINFLAG_MUTED | ADMINFLAG_SELFMUTED)) &&
@@ -3175,12 +3175,12 @@
talkreq_manager = 1;
ast_manager_event(chan, EVENT_FLAG_CALL, "MeetmeTalkRequest",
- "Channel: %s\r\n"
- "Uniqueid: %s\r\n"
- "Meetme: %s\r\n"
- "Usernum: %i\r\n"
- "Status: on\r\n",
- ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no);
+ "Channel: %s\r\n"
+ "Uniqueid: %s\r\n"
+ "Meetme: %s\r\n"
+ "Usernum: %d\r\n"
+ "Status: on\r\n",
+ ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no);
}
@@ -3188,12 +3188,12 @@
!(user->adminflags & ADMINFLAG_T_REQUEST) && (talkreq_manager)) {
talkreq_manager = 0;
ast_manager_event(chan, EVENT_FLAG_CALL, "MeetmeTalkRequest",
- "Channel: %s\r\n"
- "Uniqueid: %s\r\n"
- "Meetme: %s\r\n"
- "Usernum: %i\r\n"
- "Status: off\r\n",
- ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no);
+ "Channel: %s\r\n"
+ "Uniqueid: %s\r\n"
+ "Meetme: %s\r\n"
+ "Usernum: %d\r\n"
+ "Status: off\r\n",
+ ast_channel_name(chan), ast_channel_uniqueid(chan), conf->confno, user->user_no);
}
/* If user have been hung up, exit the conference */
@@ -4578,9 +4578,8 @@
{
struct ast_conf_user *user = NULL;
int cid;
-
- sscanf(callerident, "%30i", &cid);
- if (conf && callerident) {
+
+ if (conf && callerident && sscanf(callerident, "%30d", &cid) == 1) {
user = ao2_find(conf->usercontainer, &cid, 0);
/* reference decremented later in admin_exec */
return user;
Modified: team/file/res_xmpp/apps/app_mixmonitor.c
URL: http://svnview.digium.com/svn/asterisk/team/file/res_xmpp/apps/app_mixmonitor.c?view=diff&rev=368711&r1=368710&r2=368711
==============================================================================
--- team/file/res_xmpp/apps/app_mixmonitor.c (original)
+++ team/file/res_xmpp/apps/app_mixmonitor.c Fri Jun 8 13:29:09 2012
@@ -42,6 +42,7 @@
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/paths.h" /* use ast_config_AST_MONITOR_DIR */
+#include "asterisk/stringfields.h"
#include "asterisk/file.h"
#include "asterisk/audiohook.h"
#include "asterisk/pbx.h"
@@ -51,6 +52,7 @@
#include "asterisk/channel.h"
#include "asterisk/autochan.h"
#include "asterisk/manager.h"
+#include "asterisk/callerid.h"
#include "asterisk/mod_format.h"
#include "asterisk/linkedlists.h"
@@ -111,6 +113,12 @@
<option name="i">
<argument name="chanvar" required="true" />
<para>Stores the MixMonitor's ID on this channel variable.</para>
+ </option>
+ <option name="m">
+ <argument name="mailbox" required="true" />
+ <para>Create a copy of the recording as a voicemail in the indicated <emphasis>mailbox</emphasis>(es)
+ separated by commas eg. m(1111 at default,2222 at default,...). Folders can be optionally specified using
+ the syntax: mailbox at context/folder</para>
</option>
</optionlist>
</parameter>
@@ -238,6 +246,17 @@
static const char * const mixmonitor_spy_type = "MixMonitor";
+/*!
+ * \internal
+ * \brief This struct is a list item holds data needed to find a vm_recipient within voicemail
+ */
+struct vm_recipient {
+ char mailbox[AST_MAX_CONTEXT];
+ char context[AST_MAX_EXTENSION];
+ char folder[80];
+ AST_LIST_ENTRY(vm_recipient) list;
+};
+
struct mixmonitor {
struct ast_audiohook audiohook;
struct ast_callid *callid;
@@ -249,6 +268,20 @@
unsigned int flags;
struct ast_autochan *autochan;
struct mixmonitor_ds *mixmonitor_ds;
+
+ /* the below string fields describe data used for creating voicemails from the recording */
+ AST_DECLARE_STRING_FIELDS(
+ AST_STRING_FIELD(call_context);
+ AST_STRING_FIELD(call_macrocontext);
+ AST_STRING_FIELD(call_extension);
+ AST_STRING_FIELD(call_callerchan);
+ AST_STRING_FIELD(call_callerid);
+ );
+ int call_priority;
+
+ /* FUTURE DEVELOPMENT NOTICE
+ * recipient_list will need locks if we make it editable after the monitor is started */
+ AST_LIST_HEAD_NOLOCK(, vm_recipient) recipient_list;
};
enum mixmonitor_flags {
@@ -260,7 +293,8 @@
MUXFLAG_READ = (1 << 6),
MUXFLAG_WRITE = (1 << 7),
MUXFLAG_COMBINED = (1 << 8),
- MUXFLAG_UID = (1 << 9),
+ MUXFLAG_UID = (1 << 9),
+ MUXFLAG_VMRECIPIENTS = (1 << 10),
};
enum mixmonitor_args {
@@ -269,7 +303,8 @@
OPT_ARG_VOLUME,
OPT_ARG_WRITENAME,
OPT_ARG_READNAME,
- OPT_ARG_UID,
+ OPT_ARG_UID,
+ OPT_ARG_VMRECIPIENTS,
OPT_ARG_ARRAY_SIZE, /* Always last element of the enum */
};
@@ -282,6 +317,7 @@
AST_APP_OPTION_ARG('r', MUXFLAG_READ, OPT_ARG_READNAME),
AST_APP_OPTION_ARG('t', MUXFLAG_WRITE, OPT_ARG_WRITENAME),
AST_APP_OPTION_ARG('i', MUXFLAG_UID, OPT_ARG_UID),
+ AST_APP_OPTION_ARG('m', MUXFLAG_VMRECIPIENTS, OPT_ARG_VMRECIPIENTS),
});
struct mixmonitor_ds {
@@ -380,6 +416,70 @@
ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);
return res;
+}
+
+/*!
+ * \internal
+ * \brief adds recipients to a mixmonitor's recipient list
+ * \param mixmonitor mixmonitor being affected
+ * \param vm_recipients string containing the desired recipients to add
+ */
+static void add_vm_recipients_from_string(struct mixmonitor *mixmonitor, const char *vm_recipients)
+{
+ /* recipients are in a single string with a format format resembling "mailbox at context/INBOX,mailbox2 at context2,mailbox3 at context3/Work" */
+ char *cur_mailbox = ast_strdupa(vm_recipients);
+ char *cur_context;
+ char *cur_folder;
+ char *next;
+ int elements_processed = 0;
+
+ while (!ast_strlen_zero(cur_mailbox)) {
+ ast_debug(3, "attempting to add next element %d from %s\n", elements_processed, cur_mailbox);
+ if ((next = strchr(cur_mailbox, ',')) || (next = strchr(cur_mailbox, '&'))) {
+ *(next++) = '\0';
+ }
+
+ if ((cur_folder = strchr(cur_mailbox, '/'))) {
+ *(cur_folder++) = '\0';
+ } else {
+ cur_folder = "INBOX";
+ }
+
+ if ((cur_context = strchr(cur_mailbox, '@'))) {
+ *(cur_context++) = '\0';
+ } else {
+ cur_context = "default";
+ }
+
+ if (!ast_strlen_zero(cur_mailbox) && !ast_strlen_zero(cur_context)) {
+ struct vm_recipient *recipient;
+ if (!(recipient = ast_malloc(sizeof(*recipient)))) {
+ ast_log(LOG_ERROR, "Failed to allocate recipient. Aborting function.\n");
+ return;
+ }
+ ast_copy_string(recipient->context, cur_context, sizeof(recipient->context));
+ ast_copy_string(recipient->mailbox, cur_mailbox, sizeof(recipient->mailbox));
+ ast_copy_string(recipient->folder, cur_folder, sizeof(recipient->folder));
+
+ /* Add to list */
+ ast_verb(5, "Adding %s@%s to recipient list\n", recipient->mailbox, recipient->context);
+ AST_LIST_INSERT_HEAD(&mixmonitor->recipient_list, recipient, list);
+ } else {
+ ast_log(LOG_ERROR, "Failed to properly parse extension and/or context from element %d of recipient string: %s\n", elements_processed, vm_recipients);
+ }
+
+ cur_mailbox = next;
+ elements_processed++;
+ }
+}
+
+static void clear_mixmonitor_recipient_list(struct mixmonitor *mixmonitor)
+{
+ struct vm_recipient *current;
+ while ((current = AST_LIST_REMOVE_HEAD(&mixmonitor->recipient_list, list))) {
+ /* Clear list element data */
+ ast_free(current);
+ }
}
#define SAMPLES_PER_FRAME 160
@@ -397,6 +497,12 @@
ast_free(mixmonitor->post_process);
}
+ /* Free everything in the recipient list */
+ clear_mixmonitor_recipient_list(mixmonitor);
+
+ /* clean stringfields */
+ ast_string_field_free_memory(mixmonitor);
+
if (mixmonitor->callid) {
ast_callid_unref(mixmonitor->callid);
}
@@ -404,10 +510,50 @@
}
}
-static void mixmonitor_save_prep(struct mixmonitor *mixmonitor, char *filename, struct ast_filestream **fs, unsigned int *oflags, int *errflag)
+/*!
+ * \internal
+ * \brief Copies the mixmonitor to all voicemail recipients
+ * \param mixmonitor The mixmonitor that needs to forward its file to recipients
+ * \param ext Format of the file that was saved
+ */
+static void copy_to_voicemail(struct mixmonitor *mixmonitor, const char *ext, const char *filename)
+{
+ struct vm_recipient *recipient = NULL;
+ struct ast_vm_recording_data recording_data;
+ if (ast_string_field_init(&recording_data, 512)) {
+ ast_log(LOG_ERROR, "Failed to string_field_init, skipping copy_to_voicemail\n");
+ return;
+ }
+
+ /* Copy strings to stringfields that will be used for all recipients */
+ ast_string_field_set(&recording_data, recording_file, filename);
+ ast_string_field_set(&recording_data, recording_ext, ext);
+ ast_string_field_set(&recording_data, call_context, mixmonitor->call_context);
+ ast_string_field_set(&recording_data, call_macrocontext, mixmonitor->call_macrocontext);
+ ast_string_field_set(&recording_data, call_extension, mixmonitor->call_extension);
+ ast_string_field_set(&recording_data, call_callerchan, mixmonitor->call_callerchan);
+ ast_string_field_set(&recording_data, call_callerid, mixmonitor->call_callerid);
+ /* and call_priority gets copied too */
+ recording_data.call_priority = mixmonitor->call_priority;
+
+ AST_LIST_TRAVERSE(&mixmonitor->recipient_list, recipient, list) {
+ /* context, mailbox, and folder need to be set per recipient */
+ ast_string_field_set(&recording_data, context, recipient->context);
+ ast_string_field_set(&recording_data, mailbox, recipient->mailbox);
+ ast_string_field_set(&recording_data, folder, recipient->folder);
+
+ ast_verb(4, "MixMonitor attempting to send voicemail copy to %s@%s\n", recording_data.mailbox,
+ recording_data.context);
+ ast_app_copy_recording_to_vm(&recording_data);
+ }
+
+ /* Free the string fields for recording_data before exiting the function. */
+ ast_string_field_free_memory(&recording_data);
+}
+
+static void mixmonitor_save_prep(struct mixmonitor *mixmonitor, char *filename, struct ast_filestream **fs, unsigned int *oflags, int *errflag, char **ext)
{
/* Initialize the file if not already done so */
- char *ext = NULL;
char *last_slash = NULL;
if (!ast_strlen_zero(filename)) {
if (!*fs && !*errflag && !mixmonitor->mixmonitor_ds->fs_quit) {
@@ -416,14 +562,15 @@
last_slash = strrchr(filename, '/');
- if ((ext = strrchr(filename, '.')) && (ext > last_slash)) {
- *(ext++) = '\0';
+ if ((*ext = strrchr(filename, '.')) && (*ext > last_slash)) {
+ **ext = '\0';
+ *ext = *ext + 1;
} else {
- ext = "raw";
+ *ext = "raw";
}
- if (!(*fs = ast_writefile(filename, ext, NULL, *oflags, 0, 0666))) {
- ast_log(LOG_ERROR, "Cannot open %s.%s\n", filename, ext);
+ if (!(*fs = ast_writefile(filename, *ext, NULL, *oflags, 0, 0666))) {
+ ast_log(LOG_ERROR, "Cannot open %s.%s\n", filename, *ext);
*errflag = 1;
} else {
struct ast_filestream *tmp = *fs;
@@ -436,6 +583,9 @@
static void *mixmonitor_thread(void *obj)
{
struct mixmonitor *mixmonitor = obj;
+ char *fs_ext = "";
+ char *fs_read_ext = "";
+ char *fs_write_ext = "";
struct ast_filestream **fs = NULL;
struct ast_filestream **fs_read = NULL;
@@ -457,9 +607,9 @@
fs_write = &mixmonitor->mixmonitor_ds->fs_write;
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
- mixmonitor_save_prep(mixmonitor, mixmonitor->filename, fs, &oflags, &errflag);
- mixmonitor_save_prep(mixmonitor, mixmonitor->filename_read, fs_read, &oflags, &errflag);
- mixmonitor_save_prep(mixmonitor, mixmonitor->filename_write, fs_write, &oflags, &errflag);
+ mixmonitor_save_prep(mixmonitor, mixmonitor->filename, fs, &oflags, &errflag, &fs_ext);
+ mixmonitor_save_prep(mixmonitor, mixmonitor->filename_read, fs_read, &oflags, &errflag, &fs_read_ext);
+ mixmonitor_save_prep(mixmonitor, mixmonitor->filename_write, fs_write, &oflags, &errflag, &fs_write_ext);
ast_format_set(&format_slin, ast_format_slin_by_rate(mixmonitor->mixmonitor_ds->samp_rate), 0);
@@ -554,6 +704,27 @@
}
ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
+
+ if (!AST_LIST_EMPTY(&mixmonitor->recipient_list)) {
+ if (ast_strlen_zero(fs_ext)) {
+ ast_log(LOG_ERROR, "No file extension set for Mixmonitor %s. Skipping copy to voicemail.\n",
+ mixmonitor -> name);
+ } else {
+ ast_verb(3, "Copying recordings for Mixmonitor %s to voicemail recipients\n", mixmonitor->name);
+ copy_to_voicemail(mixmonitor, fs_ext, mixmonitor->filename);
+ }
+ if (!ast_strlen_zero(fs_read_ext)) {
+ ast_verb(3, "Copying read recording for Mixmonitor %s to voicemail recipients\n", mixmonitor->name);
+ copy_to_voicemail(mixmonitor, fs_read_ext, mixmonitor->filename_read);
+ }
+ if (!ast_strlen_zero(fs_write_ext)) {
+ ast_verb(3, "Copying write recording for Mixmonitor %s to voicemail recipients\n", mixmonitor->name);
+ copy_to_voicemail(mixmonitor, fs_write_ext, mixmonitor->filename_write);
+ }
+ } else {
+ ast_debug(3, "No recipients to forward monitor to, moving on.\n");
+ }
+
mixmonitor_free(mixmonitor);
return NULL;
}
@@ -597,7 +768,8 @@
static void launch_monitor_thread(struct ast_channel *chan, const char *filename,
unsigned int flags, int readvol, int writevol,
const char *post_process, const char *filename_write,
- char *filename_read, const char *uid_channel_var)
+ char *filename_read, const char *uid_channel_var,
+ const char *recipients)
{
pthread_t thread;
struct mixmonitor *mixmonitor;
@@ -623,6 +795,12 @@
return;
}
+ /* Now that the struct has been calloced, go ahead and initialize the string fields. */
+ if (ast_string_field_init(mixmonitor, 512)) {
+ mixmonitor_free(mixmonitor);
+ return;
+ }
+
/* Setup the actual spy before creating our thread */
if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type, 0)) {
mixmonitor_free(mixmonitor);
@@ -650,7 +828,6 @@
}
ast_free(datastore_id);
-
mixmonitor->name = ast_strdup(ast_channel_name(chan));
if (!ast_strlen_zero(postprocess2)) {
@@ -667,6 +844,35 @@
if (!ast_strlen_zero(filename_read)) {
mixmonitor->filename_read = ast_strdup(filename_read);
+ }
+
+ if (!ast_strlen_zero(recipients)) {
+ char callerid[256];
+ struct ast_party_connected_line *connected;
+
+ ast_channel_lock(chan);
+
+ /* We use the connected line of the invoking channel for caller ID. */
+
+ connected = ast_channel_connected(chan);
+ ast_debug(3, "Connected Line CID = %d - %s : %d - %s\n", connected->id.name.valid,
+ connected->id.name.str, connected->id.number.valid,
+ connected->id.number.str);
+ ast_callerid_merge(callerid, sizeof(callerid),
+ S_COR(connected->id.name.valid, connected->id.name.str, NULL),
+ S_COR(connected->id.number.valid, connected->id.number.str, NULL),
+ "Unknown");
+
+ ast_string_field_set(mixmonitor, call_context, ast_channel_context(chan));
+ ast_string_field_set(mixmonitor, call_macrocontext, ast_channel_macrocontext(chan));
+ ast_string_field_set(mixmonitor, call_extension, ast_channel_exten(chan));
+ ast_string_field_set(mixmonitor, call_callerchan, ast_channel_name(chan));
+ ast_string_field_set(mixmonitor, call_callerid, callerid);
+ mixmonitor->call_priority = ast_channel_priority(chan);
+
+ ast_channel_unlock(chan);
+
+ add_vm_recipients_from_string(mixmonitor, recipients);
}
ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);
@@ -723,6 +929,7 @@
char *uid_channel_var = NULL;
struct ast_flags flags = { 0 };
+ char *recipients = NULL;
char *parse;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(filename);
@@ -774,6 +981,14 @@
}
}
+ if (ast_test_flag(&flags, MUXFLAG_VMRECIPIENTS)) {
+ if (ast_strlen_zero(opts[OPT_ARG_VMRECIPIENTS])) {
+ ast_log(LOG_WARNING, "No voicemail recipients were specified for the vm copy ('m') option.\n");
+ } else {
+ recipients = ast_strdupa(opts[OPT_ARG_VMRECIPIENTS]);
+ }
+ }
+
if (ast_test_flag(&flags, MUXFLAG_WRITE)) {
filename_write = ast_strdupa(filename_parse(opts[OPT_ARG_WRITENAME], filename_buffer, sizeof(filename_buffer)));
}
@@ -799,7 +1014,16 @@
}
pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
- launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process, filename_write, filename_read, uid_channel_var);
+ launch_monitor_thread(chan,
+ args.filename,
+ flags.flags,
+ readvol,
+ writevol,
+ args.post_process,
+ filename_write,
+ filename_read,
+ uid_channel_var,
+ recipients);
return 0;
}
Modified: team/file/res_xmpp/apps/app_queue.c
URL: http://svnview.digium.com/svn/asterisk/team/file/res_xmpp/apps/app_queue.c?view=diff&rev=368711&r1=368710&r2=368711
==============================================================================
--- team/file/res_xmpp/apps/app_queue.c (original)
+++ team/file/res_xmpp/apps/app_queue.c Fri Jun 8 13:29:09 2012
@@ -1703,12 +1703,18 @@
return state;
}
-static int extension_state_cb(const char *context, const char *exten, enum ast_extension_states state, void *data)
+static int extension_state_cb(char *context, char *exten, struct ast_state_cb_info *info, void *data)
{
struct ao2_iterator miter, qiter;
struct member *m;
struct call_queue *q;
+ int state = info->exten_state;
int found = 0, device_state = extensionstate2devicestate(state);
+
+ /* only interested in extension state updates involving device states */
+ if (info->reason != AST_HINT_UPDATE_DEVICE) {
+ return 0;
+ }
qiter = ao2_iterator_init(queues, 0);
while ((q = ao2_t_iterator_next(&qiter, "Iterate through queues"))) {
@@ -2721,11 +2727,11 @@
*/
if (!inserted && (qe->prio >= cur->prio) && position && (position <= pos + 1)) {
insert_entry(q, prev, qe, &pos);
+ inserted = 1;
/*pos is incremented inside insert_entry, so don't need to add 1 here*/
if (position < pos) {
ast_log(LOG_NOTICE, "Asked to be inserted at position %d but forced into position %d due to higher priority callers\n", position, pos);
}
- inserted = 1;
}
cur->pos = ++pos;
prev = cur;
@@ -3097,7 +3103,7 @@
/* Hangup any existing lines we have open */
if (outgoing->chan && (outgoing->chan != exception)) {
if (exception || cancel_answered_elsewhere) {
- ast_set_flag(ast_channel_flags(outgoing->chan), AST_FLAG_ANSWERED_ELSEWHERE);
+ ast_channel_hangupcause_set(outgoing->chan, AST_CAUSE_ANSWERED_ELSEWHERE);
}
ast_hangup(outgoing->chan);
}
@@ -3294,7 +3300,7 @@
return 0;
}
- if (tmp->member->ringinuse) {
+ if (!tmp->member->ringinuse) {
if (check_state_unknown && (tmp->member->status == AST_DEVICE_UNKNOWN)) {
newstate = ast_device_state(tmp->member->interface);
if (newstate != tmp->member->status) {
@@ -3351,7 +3357,7 @@
ast_channel_lock_both(tmp->chan, qe->chan);
if (qe->cancel_answered_elsewhere) {
- ast_set_flag(ast_channel_flags(tmp->chan), AST_FLAG_ANSWERED_ELSEWHERE);
+ ast_channel_hangupcause_set(tmp->chan, AST_CAUSE_ANSWERED_ELSEWHERE);
}
ast_channel_appl_set(tmp->chan, "AppQueue");
ast_channel_data_set(tmp->chan, "(Outgoing Line)");
@@ -4819,10 +4825,10 @@
qe->cancel_answered_elsewhere = 1;
}
- /* if the calling channel has the ANSWERED_ELSEWHERE flag set, make sure this is inherited.
+ /* if the calling channel has AST_CAUSE_ANSWERED_ELSEWHERE set, make sure this is inherited.
(this is mainly to support chan_local)
*/
- if (ast_test_flag(ast_channel_flags(qe->chan), AST_FLAG_ANSWERED_ELSEWHERE)) {
+ if (ast_channel_hangupcause(qe->chan) == AST_CAUSE_ANSWERED_ELSEWHERE) {
qe->cancel_answered_elsewhere = 1;
}
@@ -5851,7 +5857,7 @@
/*!
* \internal
* \brief Sets members penalty, if queuename=NULL we set member penalty in all the queues.
- * \param[in] queuename If specified, only act on a mem`ber if it belongs to this queue
+ * \param[in] queuename If specified, only act on a member if it belongs to this queue
* \param[in] interface Interface of queue member(s) having priority set.
* \param[in] property Which queue property is being set
* \param[in] penalty Value penalty is being changed to for each member
@@ -6374,6 +6380,18 @@
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
+ ast_debug(1, "queue: %s, options: %s, url: %s, announce: %s, timeout: %s, agi: %s, macro: %s, gosub: %s, rule: %s, position: %s\n",
+ args.queuename,
+ S_OR(args.options, ""),
+ S_OR(args.url, ""),
+ S_OR(args.announceoverride, ""),
+ S_OR(args.queuetimeoutstr, ""),
+ S_OR(args.agi, ""),
+ S_OR(args.macro, ""),
+ S_OR(args.gosub, ""),
+ S_OR(args.rule, ""),
+ S_OR(args.position, ""));
+
if (!ast_strlen_zero(args.options)) {
ast_app_parse_options(queue_exec_options, &opts, opt_args, args.options);
}
@@ -6451,8 +6469,8 @@
}
}
- ast_debug(1, "queue: %s, options: %s, url: %s, announce: %s, expires: %ld, priority: %d\n",
- args.queuename, args.options, args.url, args.announceoverride, (long)qe.expire, prio);
+ ast_debug(1, "queue: %s, expires: %ld, priority: %d\n",
+ args.queuename, (long)qe.expire, prio);
qe.chan = chan;
qe.prio = prio;
@@ -6468,6 +6486,8 @@
set_queue_result(chan, reason);
return 0;
}
+ ast_assert(qe.parent != NULL);
+
ast_queue_log(args.queuename, ast_channel_uniqueid(chan), "NONE", "ENTERQUEUE", "%s|%s|%d",
S_OR(args.url, ""),
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, ""),
@@ -6625,12 +6645,13 @@
if (reason != QUEUE_UNKNOWN)
set_queue_result(chan, reason);
- if (qe.parent) {
- /* every queue_ent is given a reference to it's parent call_queue when it joins the queue.
- * This ref must be taken away right before the queue_ent is destroyed. In this case
- * the queue_ent is about to be returned on the stack */
- qe.parent = queue_unref(qe.parent);
- }
+ /*
+ * every queue_ent is given a reference to it's parent
+ * call_queue when it joins the queue. This ref must be taken
+ * away right before the queue_ent is destroyed. In this case
+ * the queue_ent is about to be returned on the stack
+ */
+ qe.parent = queue_unref(qe.parent);
return res;
}
Modified: team/file/res_xmpp/apps/app_skel.c
URL: http://svnview.digium.com/svn/asterisk/team/file/res_xmpp/apps/app_skel.c?view=diff&rev=368711&r1=368710&r2=368711
==============================================================================
--- team/file/res_xmpp/apps/app_skel.c (original)
+++ team/file/res_xmpp/apps/app_skel.c Fri Jun 8 13:29:09 2012
@@ -15,8 +15,8 @@
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*
- * Please follow coding guidelines
- * http://svn.digium.com/view/asterisk/trunk/doc/CODING-GUIDELINES
+ * Please follow coding guidelines
+ * https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines
*/
/*! \file
@@ -24,8 +24,8 @@
* \brief Skeleton application
*
* \author\verbatim <Your Name Here> <<Your Email Here>> \endverbatim
- *
- * This is a skeleton for development of an Asterisk application
+ *
+ * This is a skeleton for development of an Asterisk application
* \ingroup applications
*/
@@ -38,79 +38,306 @@
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+#include <math.h> /* log10 */
#include "asterisk/file.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/lock.h"
#include "asterisk/app.h"
+#include "asterisk/config.h"
+#include "asterisk/config_options.h"
+#include "asterisk/say.h"
+#include "asterisk/astobj2.h"
+#include "asterisk/acl.h"
+#include "asterisk/netsock2.h"
+#include "asterisk/strings.h"
+#include "asterisk/cli.h"
/*** DOCUMENTATION
- <application name="Skel" language="en_US">
+ <application name="SkelGuessNumber" language="en_US">
<synopsis>
- Simple one line explaination.
+ An example number guessing game
</synopsis>
<syntax>
- <parameter name="dummy" required="true"/>
+ <parameter name="level" required="true"/>
<parameter name="options">
<optionlist>
- <option name="a">
- <para>Option A.</para>
+ <option name="c">
+ <para>The computer should cheat</para>
</option>
- <option name="b">
- <para>Option B.</para>
- </option>
- <option name="c">
- <para>Option C.</para>
+ <option name="n">
+ <para>How many games to play before hanging up</para>
</option>
</optionlist>
</parameter>
</syntax>
<description>
- <para>This application is a template to build other applications from.
- It shows you the basic structure to create your own Asterisk applications.</para>
+ <para>This simple number guessing application is a template to build other applications
+ from. It shows you the basic structure to create your own Asterisk applications.</para>
</description>
</application>
***/
-static char *app = "Skel";
+static char *app = "SkelGuessNumber";
enum option_flags {
- OPTION_A = (1 << 0),
- OPTION_B = (1 << 1),
- OPTION_C = (1 << 2),
+ OPTION_CHEAT = (1 << 0),
+ OPTION_NUMGAMES = (1 << 1),
};
enum option_args {
- OPTION_ARG_B = 0,
- OPTION_ARG_C = 1,
+ OPTION_ARG_NUMGAMES,
/* This *must* be the last value in this enum! */
- OPTION_ARG_ARRAY_SIZE = 2,
+ OPTION_ARG_ARRAY_SIZE,
};
AST_APP_OPTIONS(app_opts,{
- AST_APP_OPTION('a', OPTION_A),
- AST_APP_OPTION_ARG('b', OPTION_B, OPTION_ARG_B),
- AST_APP_OPTION_ARG('c', OPTION_C, OPTION_ARG_C),
+ AST_APP_OPTION('c', OPTION_CHEAT),
+ AST_APP_OPTION_ARG('n', OPTION_NUMGAMES, OPTION_ARG_NUMGAMES),
});
+/*! \brief A structure to hold global configuration-related options */
+struct skel_global_config {
+ AST_DECLARE_STRING_FIELDS(
+ AST_STRING_FIELD(prompt); /*!< The comma-separated list of sounds to prompt to enter a number */
+ AST_STRING_FIELD(wrong); /*!< The comma-separated list of sounds to indicate a wrong guess */
+ AST_STRING_FIELD(right); /*!< The comma-separated list of sounds to indicate a right guess */
+ AST_STRING_FIELD(high); /*!< The comma-separated list of sounds to indicate a high guess */
+ AST_STRING_FIELD(low); /*!< The comma-separated list of sounds to indicate a low guess */
+ AST_STRING_FIELD(lose); /*!< The comma-separated list of sounds to indicate a lost game */
+ );
+ uint32_t num_games; /*!< The number of games to play before hanging up */
+ unsigned char cheat:1; /*!< Whether the computer can cheat or not */
+};
+
[... 9798 lines stripped ...]
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