[svn-commits] oej: branch oej/darjeeling-prack-1.8 r402875 - in /team/oej/darjeeling-prack-...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Nov 18 07:16:53 CST 2013
Author: oej
Date: Mon Nov 18 07:16:37 2013
New Revision: 402875
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=402875
Log:
Resetting
Modified:
team/oej/darjeeling-prack-1.8/ (props changed)
team/oej/darjeeling-prack-1.8/Makefile
team/oej/darjeeling-prack-1.8/UPGRADE.txt
team/oej/darjeeling-prack-1.8/apps/app_meetme.c
team/oej/darjeeling-prack-1.8/apps/app_queue.c
team/oej/darjeeling-prack-1.8/apps/app_voicemail.c
team/oej/darjeeling-prack-1.8/build_tools/prep_tarball
team/oej/darjeeling-prack-1.8/cdr/cdr_adaptive_odbc.c
team/oej/darjeeling-prack-1.8/channels/chan_dahdi.c
team/oej/darjeeling-prack-1.8/channels/chan_iax2.c
team/oej/darjeeling-prack-1.8/channels/chan_mgcp.c
team/oej/darjeeling-prack-1.8/channels/chan_sip.c
team/oej/darjeeling-prack-1.8/channels/sig_analog.c
team/oej/darjeeling-prack-1.8/channels/sig_ss7.c
team/oej/darjeeling-prack-1.8/channels/sip/include/sip.h
team/oej/darjeeling-prack-1.8/channels/sip/reqresp_parser.c
team/oej/darjeeling-prack-1.8/codecs/ilbc/doCPLC.c
team/oej/darjeeling-prack-1.8/configs/chan_dahdi.conf.sample
team/oej/darjeeling-prack-1.8/configs/sip.conf.sample
team/oej/darjeeling-prack-1.8/configure
team/oej/darjeeling-prack-1.8/configure.ac
team/oej/darjeeling-prack-1.8/funcs/func_config.c
team/oej/darjeeling-prack-1.8/funcs/func_math.c
team/oej/darjeeling-prack-1.8/include/asterisk/pbx.h
team/oej/darjeeling-prack-1.8/include/asterisk/rtp_engine.h
team/oej/darjeeling-prack-1.8/main/abstract_jb.c
team/oej/darjeeling-prack-1.8/main/app.c
team/oej/darjeeling-prack-1.8/main/asterisk.c
team/oej/darjeeling-prack-1.8/main/astobj2.c
team/oej/darjeeling-prack-1.8/main/channel.c
team/oej/darjeeling-prack-1.8/main/data.c
team/oej/darjeeling-prack-1.8/main/editline/readline.c
team/oej/darjeeling-prack-1.8/main/editline/term.c
team/oej/darjeeling-prack-1.8/main/features.c
team/oej/darjeeling-prack-1.8/main/jitterbuf.c
team/oej/darjeeling-prack-1.8/main/loader.c
team/oej/darjeeling-prack-1.8/main/pbx.c
team/oej/darjeeling-prack-1.8/main/rtp_engine.c
team/oej/darjeeling-prack-1.8/main/test.c
team/oej/darjeeling-prack-1.8/main/translate.c
team/oej/darjeeling-prack-1.8/main/utils.c
team/oej/darjeeling-prack-1.8/main/xmldoc.c
team/oej/darjeeling-prack-1.8/makeopts.in
team/oej/darjeeling-prack-1.8/res/res_jabber.c
team/oej/darjeeling-prack-1.8/res/res_musiconhold.c
team/oej/darjeeling-prack-1.8/res/res_rtp_asterisk.c
team/oej/darjeeling-prack-1.8/res/res_rtp_multicast.c
team/oej/darjeeling-prack-1.8/sounds/Makefile
team/oej/darjeeling-prack-1.8/tests/test_dlinklists.c
team/oej/darjeeling-prack-1.8/tests/test_linkedlists.c
team/oej/darjeeling-prack-1.8/utils/clicompat.c
Propchange: team/oej/darjeeling-prack-1.8/
------------------------------------------------------------------------------
automerge = Is-there-life-off-net?
Propchange: team/oej/darjeeling-prack-1.8/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Nov 18 07:16:37 2013
@@ -1,1 +1,1 @@
-/branches/1.8:1-398774
+/branches/1.8:1-402874
Modified: team/oej/darjeeling-prack-1.8/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/Makefile?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/Makefile (original)
+++ team/oej/darjeeling-prack-1.8/Makefile Mon Nov 18 07:16:37 2013
@@ -170,7 +170,7 @@
_ASTCFLAGS+=-Wall
endif
-_ASTCFLAGS+=-Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG)
+_ASTCFLAGS+=-Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(AST_NESTED_FUNCTIONS) $(DEBUG)
ifeq ($(AST_DEVMODE),yes)
_ASTCFLAGS+=-Werror
Modified: team/oej/darjeeling-prack-1.8/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/UPGRADE.txt?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/UPGRADE.txt (original)
+++ team/oej/darjeeling-prack-1.8/UPGRADE.txt Mon Nov 18 07:16:37 2013
@@ -23,6 +23,10 @@
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.
+
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.
Modified: team/oej/darjeeling-prack-1.8/apps/app_meetme.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/apps/app_meetme.c?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/apps/app_meetme.c (original)
+++ team/oej/darjeeling-prack-1.8/apps/app_meetme.c Mon Nov 18 07:16:37 2013
@@ -4838,6 +4838,23 @@
res = -2;
goto usernotfound;
}
+ } else {
+ /* fail for commands that require a user */
+ switch (*args.command) {
+ case 'm': /* Unmute */
+ case 'M': /* Mute */
+ case 't': /* Lower user's talk volume */
+ case 'T': /* Raise user's talk volume */
+ case 'u': /* Lower user's listen volume */
+ case 'U': /* Raise user's listen volume */
+ case 'r': /* Reset user's volume level */
+ case 'k': /* Kick user */
+ res = -2;
+ ast_log(LOG_NOTICE, "No user specified!\n");
+ goto usernotfound;
+ default:
+ break;
+ }
}
switch (*args.command) {
@@ -4853,21 +4870,24 @@
case 101: /* e: Eject last user*/
{
int max_no = 0;
-
- /* If they passed in a user, disregard it */
- if (user) {
- ao2_ref(user, -1);
- }
+ struct ast_conf_user *eject_user;
ao2_callback(cnf->usercontainer, OBJ_NODATA, user_max_cmp, &max_no);
- user = ao2_find(cnf->usercontainer, &max_no, 0);
- if (!ast_test_flag64(&user->userflags, CONFFLAG_ADMIN))
- user->adminflags |= ADMINFLAG_KICKME;
- else {
+ eject_user = ao2_find(cnf->usercontainer, &max_no, 0);
+ if (!eject_user) {
+ res = -1;
+ ast_log(LOG_NOTICE, "No last user to kick!\n");
+ break;
+ }
+
+ if (!ast_test_flag64(&eject_user->userflags, CONFFLAG_ADMIN)) {
+ eject_user->adminflags |= ADMINFLAG_KICKME;
+ } else {
res = -1;
ast_log(LOG_NOTICE, "Not kicking last user, is an Admin!\n");
}
- ao2_ref(user, -1);
+
+ ao2_ref(eject_user, -1);
break;
}
case 77: /* M: Mute */
Modified: team/oej/darjeeling-prack-1.8/apps/app_queue.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/apps/app_queue.c?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/apps/app_queue.c (original)
+++ team/oej/darjeeling-prack-1.8/apps/app_queue.c Mon Nov 18 07:16:37 2013
@@ -1406,7 +1406,7 @@
ao2_lock(q);
mem_iter = ao2_iterator_init(q->members, 0);
for (; (member = ao2_iterator_next(&mem_iter)); ao2_ref(member, -1)) {
- if ((max_penalty && (member->penalty > max_penalty)) || (min_penalty && (member->penalty < min_penalty))) {
+ if ((max_penalty != INT_MAX && member->penalty > max_penalty) || (min_penalty != INT_MAX && member->penalty < min_penalty)) {
if (conditions & QUEUE_EMPTY_PENALTY) {
ast_debug(4, "%s is unavailable because his penalty is not between %d and %d\n", member->membername, min_penalty, max_penalty);
continue;
@@ -4173,23 +4173,54 @@
*/
static void update_qe_rule(struct queue_ent *qe)
{
- int max_penalty = qe->pr->max_relative ? qe->max_penalty + qe->pr->max_value : qe->pr->max_value;
- int min_penalty = qe->pr->min_relative ? qe->min_penalty + qe->pr->min_value : qe->pr->min_value;
- char max_penalty_str[20], min_penalty_str[20];
- /* a relative change to the penalty could put it below 0 */
- if (max_penalty < 0)
- max_penalty = 0;
- if (min_penalty < 0)
- min_penalty = 0;
- if (min_penalty > max_penalty)
- min_penalty = max_penalty;
- snprintf(max_penalty_str, sizeof(max_penalty_str), "%d", max_penalty);
- snprintf(min_penalty_str, sizeof(min_penalty_str), "%d", min_penalty);
- pbx_builtin_setvar_helper(qe->chan, "QUEUE_MAX_PENALTY", max_penalty_str);
- pbx_builtin_setvar_helper(qe->chan, "QUEUE_MIN_PENALTY", min_penalty_str);
- qe->max_penalty = max_penalty;
- qe->min_penalty = min_penalty;
- ast_debug(3, "Setting max penalty to %d and min penalty to %d for caller %s since %d seconds have elapsed\n", qe->max_penalty, qe->min_penalty, qe->chan->name, qe->pr->time);
+ int max_penalty = INT_MAX;
+
+ if (qe->max_penalty != INT_MAX) {
+ char max_penalty_str[20];
+
+ if (qe->pr->max_relative) {
+ max_penalty = qe->max_penalty + qe->pr->max_value;
+ } else {
+ max_penalty = qe->pr->max_value;
+ }
+
+ /* a relative change to the penalty could put it below 0 */
+ if (max_penalty < 0) {
+ max_penalty = 0;
+ }
+
+ snprintf(max_penalty_str, sizeof(max_penalty_str), "%d", max_penalty);
+ pbx_builtin_setvar_helper(qe->chan, "QUEUE_MAX_PENALTY", max_penalty_str);
+ qe->max_penalty = max_penalty;
+ ast_debug(3, "Setting max penalty to %d for caller %s since %d seconds have elapsed\n",
+ qe->max_penalty, qe->chan->name, qe->pr->time);
+ }
+
+ if (qe->min_penalty != INT_MAX) {
+ char min_penalty_str[20];
+ int min_penalty;
+
+ if (qe->pr->min_relative) {
+ min_penalty = qe->min_penalty + qe->pr->min_value;
+ } else {
+ min_penalty = qe->pr->min_value;
+ }
+
+ if (min_penalty < 0) {
+ min_penalty = 0;
+ }
+
+ if (max_penalty != INT_MAX && min_penalty > max_penalty) {
+ min_penalty = max_penalty;
+ }
+
+ snprintf(min_penalty_str, sizeof(min_penalty_str), "%d", min_penalty);
+ pbx_builtin_setvar_helper(qe->chan, "QUEUE_MIN_PENALTY", min_penalty_str);
+ qe->min_penalty = min_penalty;
+ ast_debug(3, "Setting min penalty to %d for caller %s since %d seconds have elapsed\n",
+ qe->min_penalty, qe->chan->name, qe->pr->time);
+ }
+
qe->pr = AST_LIST_NEXT(qe->pr, list);
}
@@ -4334,8 +4365,8 @@
unsigned char usepenalty = (membercount <= q->penaltymemberslimit) ? 0 : 1;
if (usepenalty) {
- if ((qe->max_penalty && (mem->penalty > qe->max_penalty)) ||
- (qe->min_penalty && (mem->penalty < qe->min_penalty))) {
+ if ((qe->max_penalty != INT_MAX && mem->penalty > qe->max_penalty) ||
+ (qe->min_penalty != INT_MAX && mem->penalty < qe->min_penalty)) {
return -1;
}
} else {
@@ -6163,10 +6194,10 @@
} else {
ast_log(LOG_WARNING, "${QUEUE_MAX_PENALTY}: Invalid value (%s), channel %s.\n",
max_penalty_str, chan->name);
- max_penalty = 0;
+ max_penalty = INT_MAX;
}
} else {
- max_penalty = 0;
+ max_penalty = INT_MAX;
}
if ((min_penalty_str = pbx_builtin_getvar_helper(chan, "QUEUE_MIN_PENALTY"))) {
@@ -6175,10 +6206,10 @@
} else {
ast_log(LOG_WARNING, "${QUEUE_MIN_PENALTY}: Invalid value (%s), channel %s.\n",
min_penalty_str, chan->name);
- min_penalty = 0;
+ min_penalty = INT_MAX;
}
} else {
- min_penalty = 0;
+ min_penalty = INT_MAX;
}
ast_channel_unlock(chan);
@@ -6352,7 +6383,7 @@
}
} else if (qe.valid_digits) {
ast_queue_log(args.queuename, chan->uniqueid, "NONE", "EXITWITHKEY",
- "%s|%d", qe.digits, qe.pos);
+ "%s|%d|%d|%ld", qe.digits, qe.pos, qe.opos, (long) time(NULL) - qe.start);
}
}
@@ -7261,9 +7292,10 @@
ao2_lock(q);
/* This check is to make sure we don't print information for realtime
* queues which have been deleted from realtime but which have not yet
- * been deleted from the in-core container
+ * been deleted from the in-core container. Only do this if we're not
+ * looking for a specific queue.
*/
- if (q->realtime) {
+ if (argc < 3 && q->realtime) {
realtime_queue = load_realtime_queue(q->name);
if (!realtime_queue) {
ao2_unlock(q);
@@ -7950,8 +7982,8 @@
case CLI_INIT:
e->command = "queue add member";
e->usage =
- "Usage: queue add member <channel> to <queue> [[[penalty <penalty>] as <membername>] state_interface <interface>]\n"
- " Add a channel to a queue with optionally: a penalty, membername and a state_interface\n";
+ "Usage: queue add member <dial string> to <queue> [[[penalty <penalty>] as <membername>] state_interface <interface>]\n"
+ " Add a dial string (Such as a channel,e.g. SIP/6001) to a queue with optionally: a penalty, membername and a state_interface\n";
return NULL;
case CLI_GENERATE:
return complete_queue_add_member(a->line, a->word, a->pos, a->n);
Modified: team/oej/darjeeling-prack-1.8/apps/app_voicemail.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/apps/app_voicemail.c?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/apps/app_voicemail.c (original)
+++ team/oej/darjeeling-prack-1.8/apps/app_voicemail.c Mon Nov 18 07:16:37 2013
@@ -12758,6 +12758,7 @@
{
int i, j, res = AST_TEST_PASS, syserr;
struct ast_vm_user *vmu;
+ struct ast_vm_user svm;
struct vm_state vms;
#ifdef IMAP_STORAGE
struct ast_channel *chan = NULL;
@@ -12810,7 +12811,7 @@
}
#endif
- if (!(vmu = find_user(NULL, testcontext, testmailbox)) &&
+ if (!(vmu = find_user(&svm, testcontext, testmailbox)) &&
!(vmu = find_or_create(testcontext, testmailbox))) {
ast_test_status_update(test, "Cannot create vmu structure\n");
ast_unreplace_sigchld();
Modified: team/oej/darjeeling-prack-1.8/build_tools/prep_tarball
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/build_tools/prep_tarball?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/build_tools/prep_tarball (original)
+++ team/oej/darjeeling-prack-1.8/build_tools/prep_tarball Mon Nov 18 07:16:37 2013
@@ -19,11 +19,11 @@
cd doc
echo "Downloading the PDF and HTML documentation from the Asterisk wiki (this will take a minute) ..."
-wget https://wiki.asterisk.org/wiki/download/attachments/19005471/Asterisk-Admin-Guide-$branch.pdf
+wget https://wiki.asterisk.org/wiki/download/attachments/19005471/Asterisk-$branch-Reference.pdf
+wget https://wiki.asterisk.org/wiki/download/attachments/19005471/Asterisk-Admin-Guide.pdf
wget https://wiki.asterisk.org/wiki/download/attachments/19005471/Asterisk-Admin-Guide-$branch.html.zip
echo "Extracting HTML Admin Guide"
unzip Asterisk-Admin-Guide-$branch.html.zip
mv AST/ Asterisk-Admin-Guide/
-mv Asterisk-Admin-Guide-$branch.pdf Asterisk-Admin-Guide.pdf
rm -f Asterisk-Admin-Guide-$branch.html.zip
echo "Documentation downloaded. Goodbye!"
Modified: team/oej/darjeeling-prack-1.8/cdr/cdr_adaptive_odbc.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/cdr/cdr_adaptive_odbc.c?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/cdr/cdr_adaptive_odbc.c (original)
+++ team/oej/darjeeling-prack-1.8/cdr/cdr_adaptive_odbc.c Mon Nov 18 07:16:37 2013
@@ -683,6 +683,11 @@
continue;
}
first = 0;
+ } else if (entry->filtervalue && entry->filtervalue[0] != '\0') {
+ ast_verb(4, "CDR column '%s' was not set and does not match filter of"
+ " '%s'. Cancelling this CDR.\n",
+ entry->cdrname, entry->filtervalue);
+ goto early_release;
}
}
Modified: team/oej/darjeeling-prack-1.8/channels/chan_dahdi.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/channels/chan_dahdi.c?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/channels/chan_dahdi.c (original)
+++ team/oej/darjeeling-prack-1.8/channels/chan_dahdi.c Mon Nov 18 07:16:37 2013
@@ -718,9 +718,9 @@
struct dahdi_pvt *oprpeer; /*!< "Operator Services" peer tech_pvt ptr */
/*! \brief Amount of gain to increase during caller id */
float cid_rxgain;
- /*! \brief Rx gain set by chan_dahdi.conf */
+ /*! \brief Software Rx gain set by chan_dahdi.conf */
float rxgain;
- /*! \brief Tx gain set by chan_dahdi.conf */
+ /*! \brief Software Tx gain set by chan_dahdi.conf */
float txgain;
float txdrc; /*!< Dynamic Range Compression factor. a number between 1 and 6ish */
@@ -1747,11 +1747,8 @@
* a failure and die, and returning 2 means no event was received. */
res = read(p->subs[index].dfd, buf, sizeof(buf));
if (res < 0) {
- if (errno != ELAST) {
- ast_log(LOG_WARNING, "read returned error: %s\n", strerror(errno));
- callerid_free(p->cs);
- return -1;
- }
+ ast_log(LOG_WARNING, "read returned error: %s\n", strerror(errno));
+ return -1;
}
if (analog_p->ringt > 0) {
@@ -11709,6 +11706,11 @@
return NULL;
}
+static void monitor_pfds_clean(void *arg) {
+ struct pollfd **pfds = arg;
+ ast_free(*pfds);
+}
+
static void *do_monitor(void *data)
{
int count, res, res2, spoint, pollres=0;
@@ -11732,6 +11734,7 @@
#endif
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, NULL);
+ pthread_cleanup_push(monitor_pfds_clean, &pfds);
for (;;) {
/* Lock the interface list */
ast_mutex_lock(&iflock);
@@ -11988,6 +11991,7 @@
ast_mutex_unlock(&iflock);
}
/* Never reached */
+ pthread_cleanup_pop(1);
return NULL;
}
@@ -15690,12 +15694,87 @@
switch (cmd) {
case CLI_INIT:
- e->command = "dahdi set hwgain";
+ e->command = "dahdi set hwgain {rx|tx}";
e->usage =
"Usage: dahdi set hwgain <rx|tx> <chan#> <gain>\n"
- " Sets the hardware gain on a a given channel, overriding the\n"
- " value provided at module loadtime, whether the channel is in\n"
- " use or not. Changes take effect immediately.\n"
+ " Sets the hardware gain on a given channel. Changes take effect\n"
+ " immediately whether the channel is in use or not.\n"
+ "\n"
+ " <rx|tx> which direction do you want to change (relative to our module)\n"
+ " <chan num> is the channel number relative to the device\n"
+ " <gain> is the gain in dB (e.g. -3.5 for -3.5dB)\n"
+ "\n"
+ " Please note:\n"
+ " * This is currently the only way to set hwgain by the channel driver.\n"
+ " * hwgain is only supportable by hardware with analog ports because\n"
+ " hwgain works on the analog side of an analog-digital conversion.\n";
+ return NULL;
+ case CLI_GENERATE:
+ return NULL;
+ }
+
+ if (a->argc != 6)
+ return CLI_SHOWUSAGE;
+
+ if (!strcasecmp("rx", a->argv[3]))
+ tx = 0; /* rx */
+ else if (!strcasecmp("tx", a->argv[3]))
+ tx = 1; /* tx */
+ else
+ return CLI_SHOWUSAGE;
+
+ channel = atoi(a->argv[4]);
+ gain = atof(a->argv[5])*10.0;
+
+ ast_mutex_lock(&iflock);
+
+ for (tmp = iflist; tmp; tmp = tmp->next) {
+
+ if (tmp->channel != channel)
+ continue;
+
+ if (tmp->subs[SUB_REAL].dfd == -1)
+ break;
+
+ hwgain.newgain = gain;
+ hwgain.tx = tx;
+ if (ioctl(tmp->subs[SUB_REAL].dfd, DAHDI_SET_HWGAIN, &hwgain) < 0) {
+ ast_cli(a->fd, "Unable to set the hardware gain for channel %d: %s\n", channel, strerror(errno));
+ ast_mutex_unlock(&iflock);
+ return CLI_FAILURE;
+ }
+ ast_cli(a->fd, "hardware %s gain set to %d (%.1f dB) on channel %d\n",
+ tx ? "tx" : "rx", gain, (float)gain/10.0, channel);
+ break;
+ }
+
+ ast_mutex_unlock(&iflock);
+
+ if (tmp)
+ return CLI_SUCCESS;
+
+ ast_cli(a->fd, "Unable to find given channel %d\n", channel);
+ return CLI_FAILURE;
+
+}
+
+static char *dahdi_set_swgain(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ int channel;
+ float gain;
+ int tx;
+ int res;
+ struct dahdi_pvt *tmp = NULL;
+
+ switch (cmd) {
+ case CLI_INIT:
+ e->command = "dahdi set swgain {rx|tx}";
+ e->usage =
+ "Usage: dahdi set swgain <rx|tx> <chan#> <gain>\n"
+ " Sets the software gain on a given channel and overrides the\n"
+ " value provided at module loadtime. Changes take effect\n"
+ " immediately whether the channel is in use or not.\n"
+ "\n"
" <rx|tx> which direction do you want to change (relative to our module)\n"
" <chan num> is the channel number relative to the device\n"
" <gain> is the gain in dB (e.g. -3.5 for -3.5dB)\n";
@@ -15715,75 +15794,6 @@
return CLI_SHOWUSAGE;
channel = atoi(a->argv[4]);
- gain = atof(a->argv[5])*10.0;
-
- ast_mutex_lock(&iflock);
-
- for (tmp = iflist; tmp; tmp = tmp->next) {
-
- if (tmp->channel != channel)
- continue;
-
- if (tmp->subs[SUB_REAL].dfd == -1)
- break;
-
- hwgain.newgain = gain;
- hwgain.tx = tx;
- if (ioctl(tmp->subs[SUB_REAL].dfd, DAHDI_SET_HWGAIN, &hwgain) < 0) {
- ast_cli(a->fd, "Unable to set the hardware gain for channel %d: %s\n", channel, strerror(errno));
- ast_mutex_unlock(&iflock);
- return CLI_FAILURE;
- }
- ast_cli(a->fd, "hardware %s gain set to %d (%.1f dB) on channel %d\n",
- tx ? "tx" : "rx", gain, (float)gain/10.0, channel);
- break;
- }
-
- ast_mutex_unlock(&iflock);
-
- if (tmp)
- return CLI_SUCCESS;
-
- ast_cli(a->fd, "Unable to find given channel %d\n", channel);
- return CLI_FAILURE;
-
-}
-
-static char *dahdi_set_swgain(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
-{
- int channel;
- float gain;
- int tx;
- int res;
- struct dahdi_pvt *tmp = NULL;
-
- switch (cmd) {
- case CLI_INIT:
- e->command = "dahdi set swgain";
- e->usage =
- "Usage: dahdi set swgain <rx|tx> <chan#> <gain>\n"
- " Sets the software gain on a a given channel, overriding the\n"
- " value provided at module loadtime, whether the channel is in\n"
- " use or not. Changes take effect immediately.\n"
- " <rx|tx> which direction do you want to change (relative to our module)\n"
- " <chan num> is the channel number relative to the device\n"
- " <gain> is the gain in dB (e.g. -3.5 for -3.5dB)\n";
- return NULL;
- case CLI_GENERATE:
- return NULL;
- }
-
- if (a->argc != 6)
- return CLI_SHOWUSAGE;
-
- if (!strcasecmp("rx", a->argv[3]))
- tx = 0; /* rx */
- else if (!strcasecmp("tx", a->argv[3]))
- tx = 1; /* tx */
- else
- return CLI_SHOWUSAGE;
-
- channel = atoi(a->argv[4]);
gain = atof(a->argv[5]);
ast_mutex_lock(&iflock);
@@ -15808,6 +15818,12 @@
ast_cli(a->fd, "software %s gain set to %.1f on channel %d\n",
tx ? "tx" : "rx", gain, channel);
+
+ if (tx) {
+ tmp->txgain = gain;
+ } else {
+ tmp->rxgain = gain;
+ }
break;
}
ast_mutex_unlock(&iflock);
@@ -16722,8 +16738,10 @@
#ifdef HAVE_PRI
for (i = 0; i < NUM_SPANS; i++) {
- if (pris[i].pri.master != AST_PTHREADT_NULL)
+ if (pris[i].pri.master != AST_PTHREADT_NULL) {
pthread_cancel(pris[i].pri.master);
+ pthread_kill(pris[i].pri.master, SIGURG);
+ }
}
ast_cli_unregister_multiple(dahdi_pri_cli, ARRAY_LEN(dahdi_pri_cli));
ast_unregister_application(dahdi_send_keypad_facility_app);
@@ -16733,9 +16751,11 @@
#endif
#if defined(HAVE_SS7)
for (i = 0; i < NUM_SPANS; i++) {
- if (linksets[i].ss7.master != AST_PTHREADT_NULL)
+ if (linksets[i].ss7.master != AST_PTHREADT_NULL) {
pthread_cancel(linksets[i].ss7.master);
- }
+ pthread_kill(linksets[i].ss7.master, SIGURG);
+ }
+ }
ast_cli_unregister_multiple(dahdi_ss7_cli, ARRAY_LEN(dahdi_ss7_cli));
#endif /* defined(HAVE_SS7) */
#if defined(HAVE_OPENR2)
@@ -16776,8 +16796,9 @@
#if defined(HAVE_PRI)
for (i = 0; i < NUM_SPANS; i++) {
- if (pris[i].pri.master && (pris[i].pri.master != AST_PTHREADT_NULL))
+ if (pris[i].pri.master && (pris[i].pri.master != AST_PTHREADT_NULL)) {
pthread_join(pris[i].pri.master, NULL);
+ }
for (j = 0; j < SIG_PRI_NUM_DCHANS; j++) {
dahdi_close_pri_fd(&(pris[i]), j);
}
@@ -16792,8 +16813,9 @@
#if defined(HAVE_SS7)
for (i = 0; i < NUM_SPANS; i++) {
- if (linksets[i].ss7.master && (linksets[i].ss7.master != AST_PTHREADT_NULL))
+ if (linksets[i].ss7.master && (linksets[i].ss7.master != AST_PTHREADT_NULL)) {
pthread_join(linksets[i].ss7.master, NULL);
+ }
for (j = 0; j < SIG_SS7_NUM_DCHANS; j++) {
dahdi_close_ss7_fd(&(linksets[i]), j);
}
Modified: team/oej/darjeeling-prack-1.8/channels/chan_iax2.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/channels/chan_iax2.c?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/channels/chan_iax2.c (original)
+++ team/oej/darjeeling-prack-1.8/channels/chan_iax2.c Mon Nov 18 07:16:37 2013
@@ -5593,31 +5593,42 @@
res = AST_BRIDGE_COMPLETE;
break;
}
- if ((f->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS) && (f->subclass.integer != AST_CONTROL_SRCUPDATE)) {
- *fo = f;
- *rc = who;
- res = AST_BRIDGE_COMPLETE;
- break;
- }
other = (who == c0) ? c1 : c0; /* the 'other' channel */
- if ((f->frametype == AST_FRAME_VOICE) ||
- (f->frametype == AST_FRAME_TEXT) ||
- (f->frametype == AST_FRAME_VIDEO) ||
- (f->frametype == AST_FRAME_IMAGE) ||
- (f->frametype == AST_FRAME_DTMF) ||
- (f->frametype == AST_FRAME_CONTROL)) {
+ if (f->frametype == AST_FRAME_CONTROL && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
+ switch (f->subclass.integer) {
+ case AST_CONTROL_VIDUPDATE:
+ case AST_CONTROL_SRCUPDATE:
+ case AST_CONTROL_SRCCHANGE:
+ case AST_CONTROL_T38_PARAMETERS:
+ ast_write(other, f);
+ break;
+ default:
+ *fo = f;
+ *rc = who;
+ res = AST_BRIDGE_COMPLETE;
+ break;
+ }
+ if (res == AST_BRIDGE_COMPLETE) {
+ break;
+ }
+ } else if (f->frametype == AST_FRAME_VOICE
+ || f->frametype == AST_FRAME_TEXT
+ || f->frametype == AST_FRAME_VIDEO
+ || f->frametype == AST_FRAME_IMAGE) {
+ ast_write(other, f);
+ } else if (f->frametype == AST_FRAME_DTMF) {
/* monitored dtmf take out of the bridge.
* check if we monitor the specific source.
*/
int monitored_source = (who == c0) ? AST_BRIDGE_DTMF_CHANNEL_0 : AST_BRIDGE_DTMF_CHANNEL_1;
- if (f->frametype == AST_FRAME_DTMF && (flags & monitored_source)) {
+
+ if (flags & monitored_source) {
*rc = who;
*fo = f;
res = AST_BRIDGE_COMPLETE;
/* Remove from native mode */
break;
}
- /* everything else goes to the other side */
ast_write(other, f);
}
ast_frfree(f);
@@ -8641,7 +8652,7 @@
realtime_update_peer(peer->name, &peer->addr, 0);
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: IAX2\r\nPeer: IAX2/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
/* modify entry in peercnts table as _not_ registered */
- peercnt_modify(0, 0, &peer->addr);
+ peercnt_modify((unsigned char) 0, 0, &peer->addr);
/* Reset the address */
memset(&peer->addr, 0, sizeof(peer->addr));
/* Reset expiry value */
@@ -8761,13 +8772,29 @@
}
}
+ /* treat an unspecified refresh interval as the minimum */
+ if (!refresh) {
+ refresh = min_reg_expire;
+ }
+ if (refresh > max_reg_expire) {
+ ast_log(LOG_NOTICE, "Restricting registration for peer '%s' to %d seconds (requested %d)\n",
+ p->name, max_reg_expire, refresh);
+ p->expiry = max_reg_expire;
+ } else if (refresh < min_reg_expire) {
+ ast_log(LOG_NOTICE, "Restricting registration for peer '%s' to %d seconds (requested %d)\n",
+ p->name, min_reg_expire, refresh);
+ p->expiry = min_reg_expire;
+ } else {
+ p->expiry = refresh;
+ }
+
if (ast_sockaddr_cmp(&p->addr, &sockaddr)) {
if (iax2_regfunk) {
iax2_regfunk(p->name, 1);
}
/* modify entry in peercnts table as _not_ registered */
- peercnt_modify(0, 0, &p->addr);
+ peercnt_modify((unsigned char) 0, 0, &p->addr);
/* Stash the IP address from which they registered */
ast_sockaddr_from_sin(&p->addr, sin);
@@ -8795,7 +8822,7 @@
/* modify entry in peercnts table as registered */
if (p->maxcallno) {
- peercnt_modify(1, p->maxcallno, &p->addr);
+ peercnt_modify((unsigned char) 1, p->maxcallno, &p->addr);
}
/* Make sure our call still exists, an INVAL at the right point may make it go away */
@@ -8813,20 +8840,7 @@
peer_unref(p);
}
}
- /* treat an unspecified refresh interval as the minimum */
- if (!refresh)
- refresh = min_reg_expire;
- if (refresh > max_reg_expire) {
- ast_log(LOG_NOTICE, "Restricting registration for peer '%s' to %d seconds (requested %d)\n",
- p->name, max_reg_expire, refresh);
- p->expiry = max_reg_expire;
- } else if (refresh < min_reg_expire) {
- ast_log(LOG_NOTICE, "Restricting registration for peer '%s' to %d seconds (requested %d)\n",
- p->name, min_reg_expire, refresh);
- p->expiry = min_reg_expire;
- } else {
- p->expiry = refresh;
- }
+
if (p->expiry && sin->sin_addr.s_addr) {
p->expire = iax2_sched_add(sched, (p->expiry + 10) * 1000, expire_registry, peer_ref(p));
if (p->expire == -1)
@@ -12512,7 +12526,7 @@
peer->pokefreqok = DEFAULT_FREQ_OK;
peer->pokefreqnotok = DEFAULT_FREQ_NOTOK;
peer->maxcallno = 0;
- peercnt_modify(0, 0, &peer->addr);
+ peercnt_modify((unsigned char) 0, 0, &peer->addr);
peer->calltoken_required = CALLTOKEN_DEFAULT;
ast_string_field_set(peer,context,"");
ast_string_field_set(peer,peercontext,"");
@@ -12697,7 +12711,7 @@
if (sscanf(v->value, "%10hu", &peer->maxcallno) != 1) {
ast_log(LOG_WARNING, "maxcallnumbers must be set to a valid number. %s is not valid at line %d.\n", v->value, v->lineno);
} else {
- peercnt_modify(1, peer->maxcallno, &peer->addr);
+ peercnt_modify((unsigned char) 1, peer->maxcallno, &peer->addr);
}
} else if (!strcasecmp(v->name, "requirecalltoken")) {
/* default is required unless in optional ip list */
Modified: team/oej/darjeeling-prack-1.8/channels/chan_mgcp.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/channels/chan_mgcp.c?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/channels/chan_mgcp.c (original)
+++ team/oej/darjeeling-prack-1.8/channels/chan_mgcp.c Mon Nov 18 07:16:37 2013
@@ -1958,7 +1958,7 @@
char *c;
char *a;
char host[258];
- int len;
+ int len = 0;
int portno;
format_t peercapability;
int peerNonCodecCapability;
@@ -1988,8 +1988,8 @@
ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c);
return -1;
}
- if (sscanf(m, "audio %30d RTP/AVP %n", &portno, &len) != 1) {
- ast_log(LOG_WARNING, "Unable to determine port number for RTP in '%s'\n", m);
+ if (sscanf(m, "audio %30d RTP/AVP %n", &portno, &len) != 1 || !len) {
+ ast_log(LOG_WARNING, "Malformed media stream descriptor: %s\n", m);
return -1;
}
sin.sin_family = AF_INET;
Modified: team/oej/darjeeling-prack-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/channels/chan_sip.c?view=diff&rev=402875&r1=402874&r2=402875
==============================================================================
--- team/oej/darjeeling-prack-1.8/channels/chan_sip.c (original)
+++ team/oej/darjeeling-prack-1.8/channels/chan_sip.c Mon Nov 18 07:16:37 2013
@@ -714,6 +714,7 @@
static int global_rtpkeepalive; /*!< Send RTP keepalives */
static int global_reg_timeout; /*!< Global time between attempts for outbound registrations */
static int global_regattempts_max; /*!< Registration attempts before giving up */
+static int global_reg_retry_403; /*!< Treat 403 responses to registrations as 401 responses */
static int global_shrinkcallerid; /*!< enable or disable shrinking of caller id */
static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
* call-limit to INT_MAX. When we remove the call-limit from the code, we can make it
@@ -1527,6 +1528,7 @@
static void build_callid_pvt(struct sip_pvt *pvt);
static void change_callid_pvt(struct sip_pvt *pvt, const char *callid);
static void build_callid_registry(struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain);
+static void build_localtag_registry(struct sip_registry *reg);
static void make_our_tag(struct sip_pvt *pvt);
static int add_header(struct sip_request *req, const char *var, const char *value);
static int add_header_max_forwards(struct sip_pvt *dialog, struct sip_request *req);
@@ -5805,6 +5807,7 @@
dialog->chanvars = copy_vars(peer->chanvars);
if (peer->fromdomainport)
dialog->fromdomainport = peer->fromdomainport;
+ dialog->callingpres = peer->callingpres;
return 0;
}
@@ -6916,6 +6919,7 @@
{
int res = 0;
struct sip_pvt *p = ast->tech_pvt;
+ int oldsdp = FALSE;
if (!p) {
ast_debug(1, "Asked to answer channel %s without tech pvt; ignoring\n",
@@ -6932,10 +6936,14 @@
if (ast->_state != AST_STATE_UP || ast_test_flag(&p->flags[2], SIP_PAGE3_INVITE_WAIT_FOR_PRACK)) {
try_suggested_sip_codec(p);
+ if (ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT)) {
+ oldsdp = TRUE;
+ }
+
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast->name);
ast_rtp_instance_update_source(p->rtp);
- res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE, TRUE);
+ res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, oldsdp, TRUE);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
sip_pvt_unlock(p);
@@ -8106,6 +8114,12 @@
const char *host = S_OR(fromdomain, ast_sockaddr_stringify_host_remote(ourip));
ast_string_field_build(reg, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
+}
+
+/*! \brief Build SIP From tag value for REGISTER */
+static void build_localtag_registry(struct sip_registry *reg)
+{
+ ast_string_field_build(reg, localtag, "as%08lx", ast_random());
}
/*! \brief Make our SIP dialog tag */
@@ -12003,7 +12017,7 @@
/* Prefer the audio codec we were requested to use, first, no matter what
Note that p->prefcodec can include video codecs, so mask them out
*/
- if (capability & p->prefcodec) {
+ if ((capability & p->prefcodec) & AST_FORMAT_AUDIO_MASK) {
format_t codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
@@ -12077,13 +12091,9 @@
/* Our T.38 end is */
ast_udptl_get_us(p->udptl, &udptladdr);
- /* Determine T.38 UDPTL destination */
- if (!ast_sockaddr_isnull(&p->udptlredirip)) {
- ast_sockaddr_copy(&udptldest, &p->udptlredirip);
- } else {
- ast_sockaddr_copy(&udptldest, &p->ourip);
- ast_sockaddr_set_port(&udptldest, ast_sockaddr_port(&udptladdr));
- }
+ /* We don't use directmedia for T.38, so keep the destination the same as our IP address. */
+ ast_sockaddr_copy(&udptldest, &p->ourip);
+ ast_sockaddr_set_port(&udptldest, ast_sockaddr_port(&udptladdr));
if (debug) {
ast_debug(1, "T.38 UDPTL is at %s port %d\n", ast_sockaddr_stringify_addr(&p->ourip), ast_sockaddr_port(&udptladdr));
@@ -12094,9 +12104,9 @@
ast_str_append(&m_modem, 0, "m=image %d udptl t38\r\n", ast_sockaddr_port(&udptldest));
- if (!ast_sockaddr_cmp(&udptldest, &dest)) {
+ if (ast_sockaddr_cmp(&udptldest, &dest)) {
ast_str_append(&m_modem, 0, "c=IN %s %s\r\n",
- (ast_sockaddr_is_ipv6(&dest) && !ast_sockaddr_is_ipv4_mapped(&dest)) ?
+ (ast_sockaddr_is_ipv6(&udptldest) && !ast_sockaddr_is_ipv4_mapped(&udptldest)) ?
"IP6" : "IP4", ast_sockaddr_stringify_addr_remote(&udptldest));
}
@@ -12877,7 +12887,6 @@
snprintf(buf, sizeof(buf), "%d", p->expiry);
add_header(&req, "Expires", buf);
} else if (sipmethod == SIP_PRACK) {
- /* Place holder */
/* Add headers for PRACK */
char buf[SIPBUFSIZE/2];
snprintf(buf, sizeof(buf), "%u %u %s", p->irseq, p->lastinvite, "INVITE");
@@ -13304,6 +13313,7 @@
case DIALOG_INFO_XML: /* SNOM subscribes in this format */
ast_str_append(tmp, 0, "<?xml version=\"1.0\"?>\n");
ast_str_append(tmp, 0, "<dialog-info xmlns=\"urn:ietf:params:xml:ns:dialog-info\" version=\"%u\" state=\"%s\" entity=\"%s\">\n", p->dialogver, full ? "full" : "partial", mto);
+
if ((state & AST_EXTENSION_RINGING) && sip_cfg.notifyringing) {
/* Twice the extension length should be enough for XML encoding */
char local_display[AST_MAX_EXTENSION * 2];
@@ -13329,29 +13339,51 @@
struct ast_channel *caller;
if ((caller = ast_channel_callback(find_calling_channel, NULL, p, 0))) {
+ static char *anonymous = "anonymous";
+ static char *invalid = "anonymous.invalid";
char *cid_num;
char *connected_num;
int need;
+ int cid_num_restricted, connected_num_restricted;
ast_channel_lock(caller);
+
+ cid_num_restricted = (caller->caller.id.number.presentation &
+ AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED;
cid_num = S_COR(caller->caller.id.number.valid,
- caller->caller.id.number.str, "");
- need = strlen(cid_num) + strlen(p->fromdomain) + sizeof("sip:@");
+ S_COR(cid_num_restricted, anonymous,
+ caller->caller.id.number.str), "");
+
+ need = strlen(cid_num) + (cid_num_restricted ? strlen(invalid) :
+ strlen(p->fromdomain)) + sizeof("sip:@");
+
remote_target = ast_alloca(need);
- snprintf(remote_target, need, "sip:%s@%s", cid_num, p->fromdomain);
+ snprintf(remote_target, need, "sip:%s@%s", cid_num,
+ cid_num_restricted ? invalid : p->fromdomain);
ast_xml_escape(S_COR(caller->caller.id.name.valid,
- caller->caller.id.name.str, ""),
+ S_COR((caller->caller.id.name.presentation &
+ AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED, anonymous,
+ caller->caller.id.name.str), ""),
remote_display, sizeof(remote_display));
+ connected_num_restricted = (caller->connected.id.number.presentation &
+ AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED;
connected_num = S_COR(caller->connected.id.number.valid,
- caller->connected.id.number.str, "");
- need = strlen(connected_num) + strlen(p->fromdomain) + sizeof("sip:@");
+ S_COR(connected_num_restricted, anonymous,
+ caller->connected.id.number.str), "");
+
+ need = strlen(connected_num) + (connected_num_restricted ? strlen(invalid) :
+ strlen(p->fromdomain)) + sizeof("sip:@");
local_target = ast_alloca(need);
- snprintf(local_target, need, "sip:%s@%s", connected_num, p->fromdomain);
+
+ snprintf(local_target, need, "sip:%s@%s", connected_num,
+ connected_num_restricted ? invalid : p->fromdomain);
ast_xml_escape(S_COR(caller->connected.id.name.valid,
- caller->connected.id.name.str, ""),
+ S_COR((caller->connected.id.name.presentation &
+ AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED, anonymous,
+ caller->connected.id.name.str), ""),
local_display, sizeof(local_display));
ast_channel_unlock(caller);
@@ -13940,13 +13972,13 @@
return 0;
} else {
p = dialog_ref(r->call, "getting a copy of the r->call dialog in transmit_register");
- make_our_tag(p); /* create a new local tag for every register attempt */
ast_string_field_set(p, theirtag, NULL); /* forget their old tag, so we don't match tags when getting response */
}
} else {
/* Build callid for registration if we haven't registered before */
if (!r->callid_valid) {
build_callid_registry(r, &internip, default_fromdomain);
+ build_localtag_registry(r);
r->callid_valid = TRUE;
}
/* Allocate SIP dialog for registration */
@@ -13954,6 +13986,9 @@
[... 1722 lines stripped ...]
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