[svn-commits] eliel: branch group/appdocsxml r153264 - in /team/group/appdocsxml: ./ apps/ ...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Oct 31 16:08:36 CDT 2008
Author: eliel
Date: Fri Oct 31 16:08:35 2008
New Revision: 153264
URL: http://svn.digium.com/view/asterisk?view=rev&rev=153264
Log:
Fix conflicts and continue.
Modified:
team/group/appdocsxml/ (props changed)
team/group/appdocsxml/CHANGES
team/group/appdocsxml/apps/app_dial.c
team/group/appdocsxml/apps/app_followme.c
team/group/appdocsxml/apps/app_page.c
team/group/appdocsxml/apps/app_queue.c
team/group/appdocsxml/include/asterisk/channel.h
team/group/appdocsxml/include/asterisk/dial.h
team/group/appdocsxml/main/dial.c
team/group/appdocsxml/main/features.c
Propchange: team/group/appdocsxml/
------------------------------------------------------------------------------
automerge = *
Propchange: team/group/appdocsxml/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Oct 31 16:08:35 2008
@@ -1,1 +1,1 @@
-/trunk:1-153179
+/trunk:1-153260
Modified: team/group/appdocsxml/CHANGES
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/CHANGES?view=diff&rev=153264&r1=153263&r2=153264
==============================================================================
--- team/group/appdocsxml/CHANGES (original)
+++ team/group/appdocsxml/CHANGES Fri Oct 31 16:08:35 2008
@@ -685,6 +685,7 @@
WaitForRing() now takes floating pt timeout arg.
SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
* Added 's' option to Page application.
+ * Added an optional timeout argument to the Page application.
* Added 'E', 'V', and 'P' commands to ExternalIVR.
* Added 'o' and 'X' options to Chanspy.
* Added a new dialplan application, Bridge, which allows you to bridge the
Modified: team/group/appdocsxml/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_dial.c?view=diff&rev=153264&r1=153263&r2=153264
==============================================================================
--- team/group/appdocsxml/apps/app_dial.c (original)
+++ team/group/appdocsxml/apps/app_dial.c Fri Oct 31 16:08:35 2008
@@ -1485,7 +1485,6 @@
int sentringing = 0, moh = 0;
const char *outbound_group = NULL;
int result = 0;
- time_t start_time;
char *parse;
int opermode = 0;
AST_DECLARE_APP_ARGS(args,
@@ -1847,7 +1846,6 @@
}
}
- time(&start_time);
peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, &result);
/* The ast_channel_datastore_remove() function could fail here if the
@@ -1876,10 +1874,9 @@
/* almost done, although the 'else' block is 400 lines */
} else {
const char *number;
- time_t end_time, answer_time = time(NULL);
- char toast[80]; /* buffer to set variables */
strcpy(pa.status, "ANSWER");
+ pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
/* Ah ha! Someone answered within the desired timeframe. Of course after this
we will always return with -1 so that it is hung up properly after the
conversation. */
@@ -2110,11 +2107,31 @@
res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
}
}
-
+
if (res) { /* some error */
res = -1;
- end_time = time(NULL);
} else {
+ auto void end_bridge_callback(void);
+ void end_bridge_callback (void)
+ {
+ char buf[80];
+ time_t end;
+
+ time(&end);
+
+ ast_channel_lock(chan);
+ if (chan->cdr->answer.tv_sec) {
+ snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->answer.tv_sec);
+ pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
+ }
+
+ if (chan->cdr->start.tv_sec) {
+ snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->start.tv_sec);
+ pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
+ }
+ ast_channel_unlock(chan);
+ }
+
if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
@@ -2138,6 +2155,8 @@
if (ast_test_flag64(peerflags, OPT_GO_ON))
ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
+ config.end_bridge_callback = end_bridge_callback;
+
if (moh) {
moh = 0;
ast_moh_stop(chan);
@@ -2164,13 +2183,7 @@
ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
}
res = ast_bridge_call(chan, peer, &config);
- end_time = time(NULL);
- snprintf(toast, sizeof(toast), "%ld", (long)(end_time - answer_time));
- pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", toast);
- }
-
- snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
- pbx_builtin_setvar_helper(chan, "DIALEDTIME", toast);
+ }
if (res != AST_PBX_NO_HANGUP_PEER_PARKED && ast_test_flag64(&opts, OPT_PEER_H)) {
ast_log(LOG_NOTICE, "PEER context: %s; PEER exten: %s; PEER priority: %d\n",
Modified: team/group/appdocsxml/apps/app_followme.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_followme.c?view=diff&rev=153264&r1=153263&r2=153264
==============================================================================
--- team/group/appdocsxml/apps/app_followme.c (original)
+++ team/group/appdocsxml/apps/app_followme.c Fri Oct 31 16:08:35 2008
@@ -148,7 +148,6 @@
});
static int ynlongest = 0;
-static time_t start_time, answer_time, end_time;
static const char *featuredigittostr;
static int featuredigittimeout = 5000; /*!< Feature Digit Timeout */
@@ -788,7 +787,6 @@
while (nm) {
ast_debug(2, "Number %s timeout %ld\n", nm->number,nm->timeout);
- time(&start_time);
number = ast_strdupa(nm->number);
ast_debug(3, "examining %s\n", number);
@@ -964,7 +962,6 @@
int duration = 0;
struct ast_channel *caller;
struct ast_channel *outbound;
- static char toast[80];
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(followmeid);
AST_APP_ARG(options);
@@ -1067,6 +1064,27 @@
ast_stream_and_wait(chan, targs.sorryprompt, "");
res = 0;
} else {
+ auto void end_bridge_callback(void);
+ void end_bridge_callback (void)
+ {
+ char buf[80];
+ time_t end;
+
+ time(&end);
+
+ ast_channel_lock(chan);
+ if (chan->cdr->answer.tv_sec) {
+ snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->answer.tv_sec);
+ pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
+ }
+
+ if (chan->cdr->start.tv_sec) {
+ snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->start.tv_sec);
+ pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
+ }
+ ast_channel_unlock(chan);
+ }
+
caller = chan;
outbound = targs.outbound;
/* Bridge the two channels. */
@@ -1075,6 +1093,7 @@
ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
+ config.end_bridge_callback = end_bridge_callback;
ast_moh_stop(caller);
/* Be sure no generators are left on it */
@@ -1086,13 +1105,7 @@
ast_hangup(outbound);
goto outrun;
}
- time(&answer_time);
res = ast_bridge_call(caller, outbound, &config);
- time(&end_time);
- snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
- pbx_builtin_setvar_helper(caller, "DIALEDTIME", toast);
- snprintf(toast, sizeof(toast), "%ld", (long)(end_time - answer_time));
- pbx_builtin_setvar_helper(caller, "ANSWEREDTIME", toast);
if (outbound)
ast_hangup(outbound);
}
Modified: team/group/appdocsxml/apps/app_page.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_page.c?view=diff&rev=153264&r1=153263&r2=153264
==============================================================================
--- team/group/appdocsxml/apps/app_page.c (original)
+++ team/group/appdocsxml/apps/app_page.c Fri Oct 31 16:08:35 2008
@@ -51,32 +51,44 @@
</synopsis>
<syntax>
<parameter name="Technology/Resource" required="true" argsep="&">
+ <argument name="Technology/Resource" required="true">
+ <para>Specification of the device(s) to dial. These must be in the format of
+ <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
+ represents a particular channel driver, and <replaceable>Resource</replaceable> represents a resource
+ available to that particular channel driver.</para>
+ </argument>
<argument name="Technology2/Resource2" multiple="true">
<para>Optional extra devices to dial inparallel</para>
- <para>If you need more then one enter them asTechnology2/Resource2&Technology3/Resourse3&.....</para>
+ <para>If you need more then one enter them as Technology2/Resource2&
+ Technology3/Resourse3&.....</para>
</argument>
- <para>Specification of the device(s) to dial.
- These must be in the format of <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
- represents a particular channel driver, and <replaceable>Resource</replaceable> represents a resource
- available to that particular channel driver.</para>
</parameter>
- <parameter name="d" required="false">
- <para>Full duplex audio</para>
+ <parameter name="options">
+ <optionlist>
+ <option name="d">
+ <para>Full duplex audio</para>
+ </option>
+ <option name="q">
+ <para>Quiet, do not play beep to caller</para>
+ </option>
+ <option name="r">
+ <para>Record the page into a file (meetme option <literal>r</literal>)</para>
+ </option>
+ <option name="s">
+ <para>Only dial channel if devicestate says its <literal>notinuse</literal></para>
+ </option>
+ </optionlist>
</parameter>
- <parameter name="q" required="false">
- <para>Quiet, do not play beep to caller</para>
- </parameter>
- <parameter name="r" required="false">
- <para>record the page into file (meetme option <literal>r</literal>)</para>
- </parameter>
- <parameter name="s" required="false">
- <para>Only dial channel if devicestate says its <literal>notinuse</literal></para>
+ <parameter name="timeout">
+ <para>Specify the length of time that the system will attempt to connect a call.
+ After this duration, any intercom calls that have not been answered will be hung up by the
+ system.</para>
</parameter>
</syntax>
<description>
<para>Places outbound calls to the given <replaceable>technology</replaceable>/<replaceable>resource</replaceable>
and dumps them into a conference bridge as muted participants. The original
- caller is dumped into conference as a speaker and the room is
+ caller is dumped into the conference as a speaker and the room is
destroyed when the original callers leaves.</para>
</description>
</application>
@@ -101,13 +113,21 @@
static int page_exec(struct ast_channel *chan, void *data)
{
- char *options, *tech, *resource, *tmp;
+ char *tech, *resource, *tmp;
char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
struct ast_flags flags = { 0 };
unsigned int confid = ast_random();
struct ast_app *app;
int res = 0, pos = 0, i = 0;
struct ast_dial *dials[MAX_DIALS];
+ int timeout = 0;
+ char *parse;
+
+ AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(devices);
+ AST_APP_ARG(options);
+ AST_APP_ARG(timeout);
+ );
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
@@ -119,21 +139,28 @@
return -1;
};
- options = ast_strdupa(data);
+ parse = ast_strdupa(data);
+
+ AST_STANDARD_APP_ARGS(args, parse);
ast_copy_string(originator, chan->name, sizeof(originator));
- if ((tmp = strchr(originator, '-')))
+ if ((tmp = strchr(originator, '-'))) {
*tmp = '\0';
-
- tmp = strsep(&options, ",");
- if (options)
- ast_app_parse_options(page_opts, &flags, opts, options);
+ }
+
+ if (!ast_strlen_zero(args.options)) {
+ ast_app_parse_options(page_opts, &flags, opts, args.options);
+ }
+
+ if (!ast_strlen_zero(args.timeout)) {
+ timeout = atoi(args.timeout);
+ }
snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
/* Go through parsing/calling each device */
- while ((tech = strsep(&tmp, "&"))) {
+ while ((tech = strsep(&args.devices, "&"))) {
int state = 0;
struct ast_dial *dial = NULL;
@@ -167,10 +194,17 @@
}
/* Append technology and resource */
- ast_dial_append(dial, tech, resource);
+ if (ast_dial_append(dial, tech, resource) == -1) {
+ ast_log(LOG_ERROR, "Failed to add %s to outbound dial\n", tech);
+ continue;
+ }
/* Set ANSWER_EXEC as global option */
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
+
+ if (timeout) {
+ ast_dial_set_global_timeout(dial, timeout * 1000);
+ }
/* Run this dial in async mode */
ast_dial_run(dial, chan, 1);
Modified: team/group/appdocsxml/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_queue.c?view=diff&rev=153264&r1=153263&r2=153264
==============================================================================
--- team/group/appdocsxml/apps/app_queue.c (original)
+++ team/group/appdocsxml/apps/app_queue.c Fri Oct 31 16:08:35 2008
@@ -3384,6 +3384,13 @@
int callcompletedinsl;
struct ao2_iterator memi;
struct ast_datastore *datastore;
+ auto void end_bridge_callback(void);
+ void end_bridge_callback(void)
+ {
+ ao2_lock(qe->parent);
+ set_queue_variables(qe);
+ ao2_unlock(qe->parent);
+ }
ast_channel_lock(qe->chan);
datastore = ast_channel_datastore_find(qe->chan, &dialed_interface_info, NULL);
@@ -3453,6 +3460,8 @@
break;
}
+
+ bridge_config.end_bridge_callback = end_bridge_callback;
/* Hold the lock while we setup the outgoing calls */
if (use_weight)
Modified: team/group/appdocsxml/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/include/asterisk/channel.h?view=diff&rev=153264&r1=153263&r2=153264
==============================================================================
--- team/group/appdocsxml/include/asterisk/channel.h (original)
+++ team/group/appdocsxml/include/asterisk/channel.h Fri Oct 31 16:08:35 2008
@@ -584,6 +584,7 @@
const char *start_sound;
int firstpass;
unsigned int flags;
+ void (* end_bridge_callback)(void); /*!< A callback that is called after a bridge attempt */
};
struct chanmon;
Modified: team/group/appdocsxml/include/asterisk/dial.h
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/include/asterisk/dial.h?view=diff&rev=153264&r1=153263&r2=153264
==============================================================================
--- team/group/appdocsxml/include/asterisk/dial.h (original)
+++ team/group/appdocsxml/include/asterisk/dial.h Fri Oct 31 16:08:35 2008
@@ -154,7 +154,7 @@
/*! \brief Set the maximum time (globally) allowed for trying to ring phones
* \param dial The dial structure to apply the time limit to
- * \param timeout Maximum time allowed
+ * \param timeout Maximum time allowed in milliseconds
* \return nothing
*/
void ast_dial_set_global_timeout(struct ast_dial *dial, int timeout);
@@ -162,7 +162,7 @@
/*! \brief Set the maximum time (per channel) allowed for trying to ring the phone
* \param dial The dial structure the channel belongs to
* \param num Channel number to set timeout on
- * \param timeout Maximum time allowed
+ * \param timeout Maximum time allowed in milliseconds
* \return nothing
*/
void ast_dial_set_timeout(struct ast_dial *dial, int num, int timeout);
Modified: team/group/appdocsxml/main/dial.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/main/dial.c?view=diff&rev=153264&r1=153263&r2=153264
==============================================================================
--- team/group/appdocsxml/main/dial.c (original)
+++ team/group/appdocsxml/main/dial.c Fri Oct 31 16:08:35 2008
@@ -1038,7 +1038,7 @@
{
dial->timeout = timeout;
- if (dial->timeout > 0 && dial->actual_timeout > dial->timeout)
+ if (dial->timeout > 0 && (dial->actual_timeout > dial->timeout || dial->actual_timeout == -1))
dial->actual_timeout = dial->timeout;
return;
@@ -1059,7 +1059,7 @@
channel->timeout = timeout;
- if (channel->timeout > 0 && dial->actual_timeout > channel->timeout)
+ if (channel->timeout > 0 && (dial->actual_timeout > channel->timeout || dial->actual_timeout == -1))
dial->actual_timeout = channel->timeout;
return;
Modified: team/group/appdocsxml/main/features.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/main/features.c?view=diff&rev=153264&r1=153263&r2=153264
==============================================================================
--- team/group/appdocsxml/main/features.c (original)
+++ team/group/appdocsxml/main/features.c Fri Oct 31 16:08:35 2008
@@ -2178,6 +2178,7 @@
int diff;
int hasfeatures=0;
int hadfeatures=0;
+ int autoloopflag;
struct ast_option_header *aoh;
struct ast_bridge_config backup_config;
struct ast_cdr *bridge_cdr = NULL;
@@ -2438,11 +2439,16 @@
}
before_you_go:
+ if (res != AST_PBX_KEEPALIVE && config->end_bridge_callback) {
+ config->end_bridge_callback();
+ }
/* run the hangup exten on the chan object IFF it was NOT involved in a parking situation
* if it were, then chan belongs to a different thread now, and might have been hung up long
* ago.
*/
+ autoloopflag = ast_test_flag(chan, AST_FLAG_IN_AUTOLOOP);
+ ast_set_flag(chan, AST_FLAG_IN_AUTOLOOP);
if (res != AST_PBX_KEEPALIVE && !ast_test_flag(&(config->features_caller),AST_FEATURE_NO_H_EXTEN) && ast_exists_extension(chan, chan->context, "h", 1, chan->cid.cid_num)) {
struct ast_cdr *swapper;
char savelastapp[AST_MAX_EXTENSION];
@@ -2486,6 +2492,7 @@
ast_copy_string(bridge_cdr->lastapp, savelastapp, sizeof(bridge_cdr->lastapp));
ast_copy_string(bridge_cdr->lastdata, savelastdata, sizeof(bridge_cdr->lastdata));
}
+ ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
/* obey the NoCDR() wishes. -- move the DISABLED flag to the bridge CDR if it was set on the channel during the bridge... */
if (res != AST_PBX_KEEPALIVE) {
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