[svn-commits] russell: branch russell/autoservice_1.4 r89642 - in /team/russell/autoservice...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Nov 27 11:22:20 CST 2007
Author: russell
Date: Tue Nov 27 11:22:19 2007
New Revision: 89642
URL: http://svn.digium.com/view/asterisk?view=rev&rev=89642
Log:
sync with 1.4 branch
Modified:
team/russell/autoservice_1.4/ (props changed)
team/russell/autoservice_1.4/apps/app_dial.c
team/russell/autoservice_1.4/apps/app_mixmonitor.c
team/russell/autoservice_1.4/apps/app_playback.c
team/russell/autoservice_1.4/channels/chan_sip.c
team/russell/autoservice_1.4/configs/cdr.conf.sample
team/russell/autoservice_1.4/configs/sip.conf.sample
team/russell/autoservice_1.4/configs/voicemail.conf.sample
team/russell/autoservice_1.4/funcs/func_env.c
team/russell/autoservice_1.4/include/asterisk/cdr.h
team/russell/autoservice_1.4/include/asterisk/rtp.h
team/russell/autoservice_1.4/main/app.c
team/russell/autoservice_1.4/main/cdr.c
team/russell/autoservice_1.4/main/dial.c
team/russell/autoservice_1.4/main/pbx.c
team/russell/autoservice_1.4/main/rtp.c
team/russell/autoservice_1.4/pbx/pbx_config.c
team/russell/autoservice_1.4/res/res_features.c
Propchange: team/russell/autoservice_1.4/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Nov 27 11:22:19 2007
@@ -1,1 +1,1 @@
-/branches/1.4:1-89583
+/branches/1.4:1-89639
Modified: team/russell/autoservice_1.4/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/apps/app_dial.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/apps/app_dial.c (original)
+++ team/russell/autoservice_1.4/apps/app_dial.c Tue Nov 27 11:22:19 2007
@@ -57,6 +57,7 @@
#include "asterisk/app.h"
#include "asterisk/causes.h"
#include "asterisk/rtp.h"
+#include "asterisk/cdr.h"
#include "asterisk/manager.h"
#include "asterisk/privacy.h"
#include "asterisk/stringfields.h"
@@ -762,7 +763,26 @@
}
}
-
+ if (peer && !ast_cdr_log_unanswered()) {
+ /* suppress the CDR's that didn't win */
+ struct dial_localuser *o;
+ for (o = outgoing; o; o = o->next) {
+ struct ast_channel *c = o->chan;
+ if (c && c != peer && c->cdr) {
+ ast_set_flag(c->cdr, AST_CDR_FLAG_POST_DISABLED);
+ }
+ }
+ } else if (!peer && !ast_cdr_log_unanswered()) {
+ /* suppress the CDR's that didn't win */
+ struct dial_localuser *o;
+ for (o = outgoing; o; o = o->next) {
+ struct ast_channel *c = o->chan;
+ if (c && c->cdr) {
+ ast_set_flag(c->cdr, AST_CDR_FLAG_POST_DISABLED);
+ }
+ }
+ }
+
return peer;
}
Modified: team/russell/autoservice_1.4/apps/app_mixmonitor.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/apps/app_mixmonitor.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/apps/app_mixmonitor.c (original)
+++ team/russell/autoservice_1.4/apps/app_mixmonitor.c Tue Nov 27 11:22:19 2007
@@ -211,14 +211,14 @@
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "End MixMonitor Recording %s\n", mixmonitor->name);
+ if (fs)
+ ast_closestream(fs);
+
if (mixmonitor->post_process) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_2 "Executing [%s]\n", mixmonitor->post_process);
ast_safe_system(mixmonitor->post_process);
}
-
- if (fs)
- ast_closestream(fs);
free(mixmonitor);
Modified: team/russell/autoservice_1.4/apps/app_playback.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/apps/app_playback.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/apps/app_playback.c (original)
+++ team/russell/autoservice_1.4/apps/app_playback.c Tue Nov 27 11:22:19 2007
@@ -152,7 +152,6 @@
struct varshead head = { .first = NULL, .last = NULL };
struct ast_var_t *n;
- ast_log(LOG_WARNING, "string <%s> depth <%d>\n", s, depth);
if (depth++ > 10) {
ast_log(LOG_WARNING, "recursion too deep, exiting\n");
return -1;
@@ -164,7 +163,6 @@
/* scan languages same as in file.c */
if (a->language == NULL)
a->language = "en"; /* default */
- ast_log(LOG_WARNING, "try <%s> in <%s>\n", s, a->language);
lang = ast_strdupa(a->language);
for (;;) {
for (v = ast_variable_browse(say_cfg, lang); v ; v = v->next) {
@@ -190,12 +188,11 @@
s = x + 1;
if ( (x = strchr(s, ':')) )
s = x + 1;
- ast_log(LOG_WARNING, "value is <%s>\n", s);
n = ast_var_assign("SAY", s);
AST_LIST_INSERT_HEAD(&head, n, entries);
/* scan the body, one piece at a time */
- while ( ret <= 0 && (x = strsep(&rule, ",")) ) { /* exit on key */
+ while ( !ret && (x = strsep(&rule, ",")) ) { /* exit on key */
char fn[128];
const char *p, *fmt, *data; /* format and data pointers */
@@ -206,7 +203,6 @@
/* replace variables */
memset(fn, 0, sizeof(fn)); /* XXX why isn't done in pbx_substitute_variables_helper! */
pbx_substitute_variables_varshead(&head, x, fn, sizeof(fn));
- ast_log(LOG_WARNING, "doing [%s]\n", fn);
/* locate prefix and data, if any */
fmt = index(fn, ':');
@@ -245,6 +241,10 @@
strcpy(fn2 + l, data);
ret = do_say(a, fn2, options, depth);
}
+
+ if (ret) {
+ break;
+ }
}
}
ast_var_delete(n);
Modified: team/russell/autoservice_1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/channels/chan_sip.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/channels/chan_sip.c (original)
+++ team/russell/autoservice_1.4/channels/chan_sip.c Tue Nov 27 11:22:19 2007
@@ -5203,16 +5203,37 @@
continue;
} else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) == 2) {
/* We have a rtpmap to handle */
- if (debug)
- ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec);
- found_rtpmap_codecs[last_rtpmap_codec] = codec;
- last_rtpmap_codec++;
+ int found = FALSE;
+ /* We should propably check if this is an audio or video codec
+ so we know where to look */
/* Note: should really look at the 'freq' and '#chans' params too */
- ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0);
- if (p->vrtp)
- ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0);
+ if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
+ ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) {
+ if (debug)
+ ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
+ found_rtpmap_codecs[last_rtpmap_codec] = codec;
+ last_rtpmap_codec++;
+ found = TRUE;
+
+ } else if (p->vrtp) {
+ if(ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) {
+ if (debug)
+ ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
+ found_rtpmap_codecs[last_rtpmap_codec] = codec;
+ last_rtpmap_codec++;
+ found = TRUE;
+ }
+ }
+ if (!found) {
+ /* Remove this codec since it's an unknown media type for us */
+ /* XXX This is buggy since the media line for audio and video can have the
+ same numbers. We need to check as described above, but for testing this works... */
+ ast_rtp_unset_m_type(newaudiortp, codec);
+ ast_rtp_unset_m_type(newvideortp, codec);
+ if (debug)
+ ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
+ }
}
}
Modified: team/russell/autoservice_1.4/configs/cdr.conf.sample
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/configs/cdr.conf.sample?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/configs/cdr.conf.sample (original)
+++ team/russell/autoservice_1.4/configs/cdr.conf.sample Tue Nov 27 11:22:19 2007
@@ -12,6 +12,16 @@
; Define whether or not to use CDR logging. Setting this to "no" will override
; any loading of backend CDR modules. Default is "yes".
;enable=yes
+
+; Define whether or not to log unanswered calls. Setting this to "yes" will
+; report every attempt to ring a phone in dialing attempts, when it was not
+; answered. For example, if you try to dial 3 extensions, and this option is "yes",
+; you will get 3 CDR's, one for each phone that was rung. Default is "no". Some
+; find this information horribly useless. Others find it very valuable. Note, in "yes"
+; mode, you will see one CDR, with one of the call targets on one side, and the originating
+; channel on the other, and then one CDR for each channel attempted. This may seem
+; redundant, but cannot be helped.
+;unanswered = no
; Define the CDR batch mode, where instead of posting the CDR at the end of
; every call, the data will be stored in a buffer to help alleviate load on the
Modified: team/russell/autoservice_1.4/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/configs/sip.conf.sample?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/configs/sip.conf.sample (original)
+++ team/russell/autoservice_1.4/configs/sip.conf.sample Tue Nov 27 11:22:19 2007
@@ -207,6 +207,9 @@
; of a friend will now be added to and compared with
; the peer limit instead of applying two call limits,
; one for the peer and one for the user.
+ ; "sip show inuse" will only show active calls on
+ ; the peer side of a "type=friend" object if this
+ ; setting is turned on.
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
;
Modified: team/russell/autoservice_1.4/configs/voicemail.conf.sample
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/configs/voicemail.conf.sample?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/configs/voicemail.conf.sample (original)
+++ team/russell/autoservice_1.4/configs/voicemail.conf.sample Tue Nov 27 11:22:19 2007
@@ -10,7 +10,8 @@
;
[general]
-; Default formats for writing Voicemail
+; Formats for writing Voicemail. Note that when using IMAP storage for
+; voicemail, only the first format specified will be used.
;format=g723sf|wav49|wav
format=wav49|gsm|wav
;
Modified: team/russell/autoservice_1.4/funcs/func_env.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/funcs/func_env.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/funcs/func_env.c (original)
+++ team/russell/autoservice_1.4/funcs/func_env.c Tue Nov 27 11:22:19 2007
@@ -73,7 +73,7 @@
char *action;
struct stat s;
- *buf = '\0';
+ ast_copy_string(buf, "0", len);
action = strsep(&data, "|");
if (stat(data, &s)) {
Modified: team/russell/autoservice_1.4/include/asterisk/cdr.h
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/include/asterisk/cdr.h?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/include/asterisk/cdr.h (original)
+++ team/russell/autoservice_1.4/include/asterisk/cdr.h Tue Nov 27 11:22:19 2007
@@ -100,6 +100,7 @@
int ast_cdr_serialize_variables(struct ast_cdr *cdr, char *buf, size_t size, char delim, char sep, int recur);
void ast_cdr_free_vars(struct ast_cdr *cdr, int recur);
int ast_cdr_copy_vars(struct ast_cdr *to_cdr, struct ast_cdr *from_cdr);
+int ast_cdr_log_unanswered(void);
typedef int (*ast_cdrbe)(struct ast_cdr *cdr);
Modified: team/russell/autoservice_1.4/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/include/asterisk/rtp.h?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/include/asterisk/rtp.h (original)
+++ team/russell/autoservice_1.4/include/asterisk/rtp.h Tue Nov 27 11:22:19 2007
@@ -170,8 +170,14 @@
/*! \brief Copy payload types between RTP structures */
void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src);
+/*! \brief Activate payload type */
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
-void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
+
+/*! \brief clear payload type */
+void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt);
+
+/*! \brief Initiate payload type to a known MIME media type for a codec */
+int ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
char *mimeType, char *mimeSubtype,
enum ast_rtp_options options);
Modified: team/russell/autoservice_1.4/main/app.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/main/app.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/main/app.c (original)
+++ team/russell/autoservice_1.4/main/app.c Tue Nov 27 11:22:19 2007
@@ -1394,7 +1394,6 @@
s = optstr;
while (*s) {
curarg = *s++ & 0x7f; /* the array (in app.h) has 128 entries */
- ast_set_flag(flags, options[curarg].flag);
argloc = options[curarg].arg_index;
if (*s == '(') {
/* Has argument */
@@ -1411,6 +1410,8 @@
} else if (argloc) {
args[argloc - 1] = NULL;
}
+ if (!argloc || !ast_strlen_zero(args[argloc - 1]))
+ ast_set_flag(flags, options[curarg].flag);
}
return res;
Modified: team/russell/autoservice_1.4/main/cdr.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/main/cdr.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/main/cdr.c (original)
+++ team/russell/autoservice_1.4/main/cdr.c Tue Nov 27 11:22:19 2007
@@ -89,6 +89,7 @@
#define BATCH_SAFE_SHUTDOWN_DEFAULT 1
static int enabled; /*! Is the CDR subsystem enabled ? */
+static int unanswered;
static int batchmode;
static int batchsize;
static int batchtime;
@@ -101,6 +102,11 @@
AST_MUTEX_DEFINE_STATIC(cdr_pending_lock);
static ast_cond_t cdr_pending_cond;
+
+int ast_cdr_log_unanswered(void)
+{
+ return unanswered;
+}
/*! Register a CDR driver. Each registered CDR driver generates a CDR
\return 0 on success, -1 on failure
@@ -984,8 +990,6 @@
struct ast_cdr_beitem *i;
for ( ; cdr ; cdr = cdr->next) {
- if (cdr->disposition < AST_CDR_ANSWERED && (ast_strlen_zero(cdr->channel) || ast_strlen_zero(cdr->dstchannel)))
- continue; /* people don't want to see unanswered single-channel events */
chan = S_OR(cdr->channel, "<unknown>");
check_post(cdr);
if (ast_tvzero(cdr->end))
@@ -1247,6 +1251,7 @@
ast_cli(fd, "CDR logging: %s\n", enabled ? "enabled" : "disabled");
ast_cli(fd, "CDR mode: %s\n", batchmode ? "batch" : "simple");
if (enabled) {
+ ast_cli(fd, "CDR output unanswered calls: %s\n", unanswered ? "yes" : "no");
if (batchmode) {
if (batch)
cnt = batch->size;
@@ -1298,6 +1303,7 @@
{
struct ast_config *config;
const char *enabled_value;
+ const char *unanswered_value;
const char *batched_value;
const char *scheduleronly_value;
const char *batchsafeshutdown_value;
@@ -1328,6 +1334,9 @@
if ((config = ast_config_load("cdr.conf"))) {
if ((enabled_value = ast_variable_retrieve(config, "general", "enable"))) {
enabled = ast_true(enabled_value);
+ }
+ if ((unanswered_value = ast_variable_retrieve(config, "general", "unanswered"))) {
+ unanswered = ast_true(unanswered_value);
}
if ((batched_value = ast_variable_retrieve(config, "general", "batch"))) {
batchmode = ast_true(batched_value);
Modified: team/russell/autoservice_1.4/main/dial.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/main/dial.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/main/dial.c (original)
+++ team/russell/autoservice_1.4/main/dial.c Tue Nov 27 11:22:19 2007
@@ -61,6 +61,7 @@
const char *device; /*! Device being dialed */
void *options[AST_DIAL_OPTION_MAX]; /*! Channel specific options */
int cause; /*! Cause code in case of failure */
+ int is_running_app:1; /*! Is this running an application? */
struct ast_channel *owner; /*! Asterisk channel */
AST_LIST_ENTRY(ast_dial_channel) list; /*! Linked list information */
};
@@ -123,8 +124,9 @@
}
/* Application execution function for 'ANSWER_EXEC' option */
-static void answer_exec_run(struct ast_channel *chan, char *app, char *args)
-{
+static void answer_exec_run(struct ast_dial *dial, struct ast_dial_channel *dial_channel, char *app, char *args)
+{
+ struct ast_channel *chan = dial_channel->owner;
struct ast_app *ast_app = pbx_findapp(app);
/* If the application was not found, return immediately */
@@ -133,6 +135,12 @@
/* All is well... execute the application */
pbx_exec(chan, ast_app, args);
+
+ /* If another thread is not taking over hang up the channel */
+ if (dial->thread != AST_PTHREADT_STOP) {
+ ast_hangup(chan);
+ dial_channel->owner = NULL;
+ }
return;
}
@@ -510,8 +518,11 @@
channel->owner = NULL;
}
/* If ANSWER_EXEC is enabled as an option, execute application on answered channel */
- if ((channel = find_relative_dial_channel(dial, who)) && (answer_exec = FIND_RELATIVE_OPTION(dial, channel, AST_DIAL_OPTION_ANSWER_EXEC)))
- answer_exec_run(who, answer_exec->app, answer_exec->args);
+ if ((channel = find_relative_dial_channel(dial, who)) && (answer_exec = FIND_RELATIVE_OPTION(dial, channel, AST_DIAL_OPTION_ANSWER_EXEC))) {
+ channel->is_running_app = 1;
+ answer_exec_run(dial, channel, answer_exec->app, answer_exec->args);
+ channel->is_running_app = 0;
+ }
} else if (dial->state == AST_DIAL_RESULT_HANGUP) {
/* Hangup everything */
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
@@ -615,8 +626,16 @@
/* Stop the thread */
dial->thread = AST_PTHREADT_STOP;
- /* Now we signal it with SIGURG so it will break out of it's waitfor */
- pthread_kill(thread, SIGURG);
+ /* If the answered channel is running an application we have to soft hangup it, can't just poke the thread */
+ if (AST_LIST_FIRST(&dial->channels)->is_running_app) {
+ struct ast_channel *chan = AST_LIST_FIRST(&dial->channels)->owner;
+ ast_channel_lock(chan);
+ ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
+ ast_channel_unlock(chan);
+ } else {
+ /* Now we signal it with SIGURG so it will break out of it's waitfor */
+ pthread_kill(thread, SIGURG);
+ }
/* Finally wait for the thread to exit */
pthread_join(thread, NULL);
Modified: team/russell/autoservice_1.4/main/pbx.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/main/pbx.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/main/pbx.c (original)
+++ team/russell/autoservice_1.4/main/pbx.c Tue Nov 27 11:22:19 2007
@@ -2315,8 +2315,10 @@
/* helper function to set extension and priority */
static void set_ext_pri(struct ast_channel *c, const char *exten, int pri)
{
+ ast_channel_lock(c);
ast_copy_string(c->exten, exten, sizeof(c->exten));
c->priority = pri;
+ ast_channel_unlock(c);
}
/*!
Modified: team/russell/autoservice_1.4/main/rtp.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/main/rtp.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/main/rtp.c (original)
+++ team/russell/autoservice_1.4/main/rtp.c Tue Nov 27 11:22:19 2007
@@ -1640,23 +1640,36 @@
ast_mutex_unlock(&rtp->bridge_lock);
}
+/*! \brief remove setting from payload type list if the rtpmap header indicates
+ an unknown media type */
+void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt)
+{
+ ast_mutex_lock(&rtp->bridge_lock);
+ rtp->current_RTP_PT[pt].isAstFormat = 0;
+ rtp->current_RTP_PT[pt].code = 0;
+ ast_mutex_unlock(&rtp->bridge_lock);
+}
+
/*! \brief Make a note of a RTP payload type (with MIME type) that was seen in
* an SDP "a=rtpmap:" line.
+ * \return 0 if the MIME type was found and set, -1 if it wasn't found
*/
-void ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt,
+int ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt,
char *mimeType, char *mimeSubtype,
enum ast_rtp_options options)
{
unsigned int i;
+ int found = 0;
if (pt < 0 || pt > MAX_RTP_PT)
- return; /* bogus payload type */
+ return -1; /* bogus payload type */
ast_mutex_lock(&rtp->bridge_lock);
for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
strcasecmp(mimeType, mimeTypes[i].type) == 0) {
+ found = 1;
rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) &&
mimeTypes[i].payloadType.isAstFormat &&
@@ -1668,7 +1681,7 @@
ast_mutex_unlock(&rtp->bridge_lock);
- return;
+ return (found ? 0 : -1);
}
/*! \brief Return the union of all of the codecs that were set by rtp_set...() calls
Modified: team/russell/autoservice_1.4/pbx/pbx_config.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/pbx/pbx_config.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/pbx/pbx_config.c (original)
+++ team/russell/autoservice_1.4/pbx/pbx_config.c Tue Nov 27 11:22:19 2007
@@ -1472,7 +1472,7 @@
if (!app_data)
app_data="";
if (ast_add_extension(argv[4], argc == 6 ? 1 : 0, exten, iprior, NULL, cidmatch, app,
- (void *)strdup(app_data), free, registrar)) {
+ (void *)strdup(app_data), ast_free, registrar)) {
switch (errno) {
case ENOMEM:
ast_cli(fd, "Out of free memory\n");
@@ -1565,7 +1565,7 @@
if (!app_data)
app_data="";
if (ast_add_extension(argv[5], argc == 7 ? 1 : 0, exten, iprior, NULL, cidmatch, app,
- (void *)strdup(app_data), free, registrar)) {
+ (void *)strdup(app_data), ast_free, registrar)) {
switch (errno) {
case ENOMEM:
ast_cli(fd, "Out of free memory\n");
Modified: team/russell/autoservice_1.4/res/res_features.c
URL: http://svn.digium.com/view/asterisk/team/russell/autoservice_1.4/res/res_features.c?view=diff&rev=89642&r1=89641&r2=89642
==============================================================================
--- team/russell/autoservice_1.4/res/res_features.c (original)
+++ team/russell/autoservice_1.4/res/res_features.c Tue Nov 27 11:22:19 2007
@@ -1921,6 +1921,7 @@
if (error) {
ast_log(LOG_WARNING, "Failed to play courtesy tone!\n");
ast_hangup(peer);
+ ast_module_user_remove(u);
return -1;
}
} else
@@ -1930,6 +1931,7 @@
if (res < 0) {
ast_log(LOG_WARNING, "Could not make channels %s and %s compatible for bridge\n", chan->name, peer->name);
ast_hangup(peer);
+ ast_module_user_remove(u);
return -1;
}
/* This runs sorta backwards, since we give the incoming channel control, as if it
@@ -1950,6 +1952,7 @@
/* Simulate the PBX hanging up */
if (res != AST_PBX_NO_HANGUP_PEER)
ast_hangup(peer);
+ ast_module_user_remove(u);
return res;
} else {
/*! \todo XXX Play a message XXX */
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