[svn-commits] jpeeler: branch jpeeler/chan_dahdi r119069 - in /team/jpeeler/chan_dahdi: ./ ...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu May 29 14:34:14 CDT 2008
Author: jpeeler
Date: Thu May 29 14:34:13 2008
New Revision: 119069
URL: http://svn.digium.com/view/asterisk?view=rev&rev=119069
Log:
sync with trunk
Added:
team/jpeeler/chan_dahdi/channels/chan_phone.h
- copied unchanged from r119015, trunk/channels/chan_phone.h
Removed:
team/jpeeler/chan_dahdi/channels/DialTone.h
Modified:
team/jpeeler/chan_dahdi/ (props changed)
team/jpeeler/chan_dahdi/CHANGES
team/jpeeler/chan_dahdi/apps/app_chanspy.c
team/jpeeler/chan_dahdi/apps/app_forkcdr.c
team/jpeeler/chan_dahdi/apps/app_milliwatt.c
team/jpeeler/chan_dahdi/apps/app_queue.c
team/jpeeler/chan_dahdi/apps/app_voicemail.c
team/jpeeler/chan_dahdi/channels/chan_agent.c
team/jpeeler/chan_dahdi/channels/chan_alsa.c
team/jpeeler/chan_dahdi/channels/chan_iax2.c
team/jpeeler/chan_dahdi/channels/chan_jingle.c
team/jpeeler/chan_dahdi/channels/chan_mgcp.c
team/jpeeler/chan_dahdi/channels/chan_phone.c
team/jpeeler/chan_dahdi/channels/chan_sip.c
team/jpeeler/chan_dahdi/channels/chan_skinny.c
team/jpeeler/chan_dahdi/configs/sip.conf.sample
team/jpeeler/chan_dahdi/include/asterisk/compat.h
team/jpeeler/chan_dahdi/include/asterisk/jingle.h
team/jpeeler/chan_dahdi/include/asterisk/lock.h
team/jpeeler/chan_dahdi/main/asterisk.c
team/jpeeler/chan_dahdi/main/cdr.c
team/jpeeler/chan_dahdi/main/cli.c
team/jpeeler/chan_dahdi/main/features.c
team/jpeeler/chan_dahdi/main/utils.c
team/jpeeler/chan_dahdi/utils/check_expr.c
Propchange: team/jpeeler/chan_dahdi/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/jpeeler/chan_dahdi/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu May 29 14:34:13 2008
@@ -1,1 +1,1 @@
-/trunk:1-118370
+/trunk:1-119068
Modified: team/jpeeler/chan_dahdi/CHANGES
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/CHANGES?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/CHANGES (original)
+++ team/jpeeler/chan_dahdi/CHANGES Thu May 29 14:34:13 2008
@@ -97,6 +97,7 @@
the register line.
* Added support for T140 RED - redundancy in T.140 to prevent text loss due to
lost packets.
+ * Added t38pt_usertpsource option. See sip.conf.sample for details.
IAX Changes
-----------
Modified: team/jpeeler/chan_dahdi/apps/app_chanspy.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/apps/app_chanspy.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/apps/app_chanspy.c (original)
+++ team/jpeeler/chan_dahdi/apps/app_chanspy.c Thu May 29 14:34:13 2008
@@ -218,6 +218,7 @@
AST_APP_OPTION('d', OPTION_DTMF_SWITCH_MODES),
});
+int next_unique_id_to_use = 0;
struct chanspy_translation_helper {
/* spy data */
@@ -294,6 +295,7 @@
struct chanspy_ds {
struct ast_channel *chan;
+ char unique_id[20];
ast_mutex_t lock;
};
@@ -546,7 +548,7 @@
chan = chanspy_ds->chan;
ast_channel_lock(chan);
- if ((datastore = ast_channel_datastore_find(chan, &chanspy_ds_info, NULL))) {
+ if ((datastore = ast_channel_datastore_find(chan, &chanspy_ds_info, chanspy_ds->unique_id))) {
ast_channel_datastore_remove(chan, datastore);
/* chanspy_ds->chan is NULL after this call */
chanspy_ds_destroy(datastore->data);
@@ -567,7 +569,7 @@
ast_mutex_lock(&chanspy_ds->lock);
- if (!(datastore = ast_channel_datastore_alloc(&chanspy_ds_info, NULL))) {
+ if (!(datastore = ast_channel_datastore_alloc(&chanspy_ds_info, chanspy_ds->unique_id))) {
ast_mutex_unlock(&chanspy_ds->lock);
chanspy_ds = chanspy_ds_free(chanspy_ds);
ast_channel_unlock(chan);
@@ -642,6 +644,8 @@
}
ast_mutex_init(&chanspy_ds.lock);
+
+ snprintf(chanspy_ds.unique_id, sizeof(chanspy_ds.unique_id), "%d", ast_atomic_fetchadd_int(&next_unique_id_to_use, +1));
if (chan->_state != AST_STATE_UP)
ast_answer(chan);
Modified: team/jpeeler/chan_dahdi/apps/app_forkcdr.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/apps/app_forkcdr.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/apps/app_forkcdr.c (original)
+++ team/jpeeler/chan_dahdi/apps/app_forkcdr.c Thu May 29 14:34:13 2008
@@ -34,6 +34,7 @@
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/cdr.h"
+#include "asterisk/app.h"
#include "asterisk/module.h"
static char *app = "ForkCDR";
@@ -43,10 +44,51 @@
" ForkCDR([options]): Causes the Call Data Record to fork an additional\n"
"cdr record starting from the time of the fork call\n"
" Options:\n"
-" v - If the option is passed all cdr variables will be passed along also.\n";
+" a - update the answer time on the NEW CDR just after it's been inited..\n"
+" The new CDR may have been answered already, the reset that forkcdr.\n"
+" does will erase the answer time. This will bring it back, but.\n"
+" the answer time will be a copy of the fork/start time. It will.\n"
+" only do this if the initial cdr was indeed already answered..\n"
+" D - Copy the disposition forward from the old cdr, after the .\n"
+" init..\n"
+" d - Clear the dstchannel on the new CDR after reset..\n"
+" e - end the original CDR. Do this after all the necc. data.\n"
+" is copied from the original CDR to the new forked CDR..\n"
+" R - do NOT reset the new cdr..\n"
+" s(name=val) - Set the CDR var 'name' in the original CDR, with value.\n"
+" 'val'.\n"
+" v - When the new CDR is forked, it gets a copy of the vars attached\n"
+" to the current CDR. The vars attached to the original CDR are removed\n"
+" unless this option is specified.\n";
-static void ast_cdr_fork(struct ast_channel *chan)
+enum {
+ OPT_SETANS = (1 << 0),
+ OPT_SETDISP = (1 << 1),
+ OPT_RESETDEST = (1 << 2),
+ OPT_ENDCDR = (1 << 3),
+ OPT_NORESET = (1 << 4),
+ OPT_KEEPVARS = (1 << 5),
+ OPT_VARSET = (1 << 6),
+};
+
+enum {
+ OPT_ARG_VARSET = 0,
+ /* note: this entry _MUST_ be the last one in the enum */
+ OPT_ARG_ARRAY_SIZE,
+};
+
+AST_APP_OPTIONS(forkcdr_exec_options, {
+ AST_APP_OPTION('a', OPT_SETANS),
+ AST_APP_OPTION('d', OPT_SETDISP),
+ AST_APP_OPTION('D', OPT_RESETDEST),
+ AST_APP_OPTION('e', OPT_ENDCDR),
+ AST_APP_OPTION('R', OPT_NORESET),
+ AST_APP_OPTION_ARG('s', OPT_VARSET, OPT_ARG_VARSET),
+ AST_APP_OPTION('v', OPT_KEEPVARS),
+});
+
+static void ast_cdr_fork(struct ast_channel *chan, struct ast_flags optflags, char *set)
{
struct ast_cdr *cdr;
struct ast_cdr *newcdr;
@@ -61,27 +103,66 @@
return;
ast_cdr_append(cdr, newcdr);
- ast_cdr_reset(newcdr, &flags);
-
+
+ if (!ast_test_flag(&optflags, OPT_NORESET))
+ ast_cdr_reset(newcdr, &flags);
+
if (!ast_test_flag(cdr, AST_CDR_FLAG_KEEP_VARS))
ast_cdr_free_vars(cdr, 0);
+ if (!ast_strlen_zero(set)) {
+ char *varname = ast_strdupa(set), *varval;
+ varval = strchr(varname,'=');
+ if (varval) {
+ *varval = 0;
+ varval++;
+ ast_cdr_setvar(cdr, varname, varval, 0);
+ }
+ }
+
+ if (ast_test_flag(&optflags, OPT_SETANS) && !ast_tvzero(cdr->answer))
+ newcdr->answer = newcdr->start;
+
+ if (ast_test_flag(&optflags, OPT_SETDISP))
+ newcdr->disposition = cdr->disposition;
+
+ if (ast_test_flag(&optflags, OPT_RESETDEST))
+ newcdr->dstchannel[0] = 0;
+
+ if (ast_test_flag(&optflags, OPT_ENDCDR))
+ ast_cdr_end(cdr);
+
ast_set_flag(cdr, AST_CDR_FLAG_CHILD | AST_CDR_FLAG_LOCKED);
}
static int forkcdr_exec(struct ast_channel *chan, void *data)
{
int res = 0;
+ char *argcopy = NULL;
+ struct ast_flags flags = {0};
+ char *opts[OPT_ARG_ARRAY_SIZE];
+ AST_DECLARE_APP_ARGS(arglist,
+ AST_APP_ARG(options);
+ );
if (!chan->cdr) {
ast_log(LOG_WARNING, "Channel does not have a CDR\n");
return 0;
}
+ argcopy = ast_strdupa(data);
+
+ AST_STANDARD_APP_ARGS(arglist, argcopy);
+
+ if (!ast_strlen_zero(arglist.options)) {
+ ast_app_parse_options(forkcdr_exec_options, &flags, opts, arglist.options);
+ } else
+ opts[OPT_ARG_VARSET] = 0;
+
if (!ast_strlen_zero(data))
- ast_set2_flag(chan->cdr, strchr(data, 'v'), AST_CDR_FLAG_KEEP_VARS);
+ ast_set2_flag(chan->cdr, ast_test_flag(&flags, OPT_KEEPVARS), AST_CDR_FLAG_KEEP_VARS);
- ast_cdr_fork(chan);
+ ast_cdr_fork(chan, flags, opts[OPT_ARG_VARSET]);
return res;
}
Modified: team/jpeeler/chan_dahdi/apps/app_milliwatt.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/apps/app_milliwatt.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/apps/app_milliwatt.c (original)
+++ team/jpeeler/chan_dahdi/apps/app_milliwatt.c Thu May 29 14:34:13 2008
@@ -25,19 +25,28 @@
* \ingroup applications
*/
+/*** MODULEINFO
+ <depend>res_indications</depend>
+ ***/
+
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
static char *app = "Milliwatt";
-static char *synopsis = "Generate a Constant 1000Hz tone at 0dbm (mu-law)";
+static char *synopsis = "Generate a Constant 1004Hz tone at 0dbm (mu-law)";
static char *descrip =
-"Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law)\n";
+" Milliwatt([options]): Generate a Constant 1004Hz tone at 0dbm.\n"
+"Previous versions of this application generated the tone at 1000Hz. If for\n"
+"some reason you would prefer that behavior, supply the 'o' option to get the\n"
+"old behavior.\n"
+"";
static char digital_milliwatt[] = {0x1e,0x0b,0x0b,0x1e,0x9e,0x8b,0x8b,0x9e} ;
@@ -74,6 +83,7 @@
ast_log(LOG_WARNING, "Only doing %d samples (%d requested)\n", maxsamples, samples);
samples = maxsamples;
}
+
len = samples * sizeof (buf[0]);
wf.datalen = len;
wf.samples = samples;
@@ -92,33 +102,56 @@
return 0;
}
-static struct ast_generator milliwattgen =
-{
+static struct ast_generator milliwattgen = {
alloc: milliwatt_alloc,
release: milliwatt_release,
generate: milliwatt_generate,
};
-static int milliwatt_exec(struct ast_channel *chan, void *data)
+static int old_milliwatt_exec(struct ast_channel *chan)
{
-
ast_set_write_format(chan, AST_FORMAT_ULAW);
ast_set_read_format(chan, AST_FORMAT_ULAW);
-
- if (chan->_state != AST_STATE_UP)
+ if (chan->_state != AST_STATE_UP) {
ast_answer(chan);
+ }
if (ast_activate_generator(chan,&milliwattgen,"milliwatt") < 0) {
ast_log(LOG_WARNING,"Failed to activate generator on '%s'\n",chan->name);
return -1;
}
- while(!ast_safe_sleep(chan, 10000));
+ while (!ast_safe_sleep(chan, 10000))
+ ;
ast_deactivate_generator(chan);
return -1;
+}
+
+static int milliwatt_exec(struct ast_channel *chan, void *data)
+{
+ const char *options = data;
+ struct ast_app *playtones_app;
+ int res = -1;
+
+ if (!ast_strlen_zero(options) && strchr(options, 'o')) {
+ return old_milliwatt_exec(chan);
+ }
+
+ if (!(playtones_app = pbx_findapp("Playtones"))) {
+ ast_log(LOG_ERROR, "The Playtones application is required to run Milliwatt()\n");
+ return -1;
+ }
+
+ res = pbx_exec(chan, playtones_app, "1004/1000");
+
+ while (!res) {
+ res = ast_safe_sleep(chan, 10000);
+ }
+
+ return res;
}
static int unload_module(void)
Modified: team/jpeeler/chan_dahdi/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/apps/app_queue.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/apps/app_queue.c (original)
+++ team/jpeeler/chan_dahdi/apps/app_queue.c Thu May 29 14:34:13 2008
@@ -3438,11 +3438,15 @@
set_queue_variables(qe);
ao2_unlock(qe->parent);
+ ast_channel_lock(qe->chan);
+ if ((monitorfilename = pbx_builtin_getvar_helper(qe->chan, "MONITOR_FILENAME"))) {
+ monitorfilename = ast_strdupa(monitorfilename);
+ }
+ ast_channel_unlock(qe->chan);
/* Begin Monitoring */
if (qe->parent->monfmt && *qe->parent->monfmt) {
if (!qe->parent->montype) {
ast_debug(1, "Starting Monitor as requested.\n");
- monitorfilename = pbx_builtin_getvar_helper(qe->chan, "MONITOR_FILENAME");
if (pbx_builtin_getvar_helper(qe->chan, "MONITOR_EXEC") || pbx_builtin_getvar_helper(qe->chan, "MONITOR_EXEC_ARGS"))
which = qe->chan;
else
@@ -3457,75 +3461,79 @@
ast_monitor_start(which, qe->parent->monfmt, tmpid, 1, X_REC_IN | X_REC_OUT);
}
} else {
- ast_debug(1, "Starting MixMonitor as requested.\n");
- monitorfilename = pbx_builtin_getvar_helper(qe->chan, "MONITOR_FILENAME");
- if (!monitorfilename) {
- if (qe->chan->cdr)
- ast_copy_string(tmpid, qe->chan->cdr->uniqueid, sizeof(tmpid));
- else
- snprintf(tmpid, sizeof(tmpid), "chan-%lx", ast_random());
- } else {
- const char *m = monitorfilename;
- for (p = tmpid2; p < tmpid2 + sizeof(tmpid2) - 1; p++, m++) {
- switch (*m) {
- case '^':
- if (*(m + 1) == '{')
- *p = '$';
- break;
- case ',':
- *p++ = '\\';
- /* Fall through */
- default:
- *p = *m;
- }
- if (*m == '\0')
- break;
- }
- if (p == tmpid2 + sizeof(tmpid2))
- tmpid2[sizeof(tmpid2) - 1] = '\0';
-
- pbx_substitute_variables_helper(qe->chan, tmpid2, tmpid, sizeof(tmpid) - 1);
- }
-
- monitor_exec = pbx_builtin_getvar_helper(qe->chan, "MONITOR_EXEC");
- monitor_options = pbx_builtin_getvar_helper(qe->chan, "MONITOR_OPTIONS");
-
- if (monitor_exec) {
- const char *m = monitor_exec;
- for (p = meid2; p < meid2 + sizeof(meid2) - 1; p++, m++) {
- switch (*m) {
- case '^':
- if (*(m + 1) == '{')
- *p = '$';
- break;
- case ',':
- *p++ = '\\';
- /* Fall through */
- default:
- *p = *m;
- }
- if (*m == '\0')
- break;
- }
- if (p == meid2 + sizeof(meid2))
- meid2[sizeof(meid2) - 1] = '\0';
-
- pbx_substitute_variables_helper(qe->chan, meid2, meid, sizeof(meid) - 1);
- }
-
- snprintf(tmpid2, sizeof(tmpid2), "%s.%s", tmpid, qe->parent->monfmt);
-
mixmonapp = pbx_findapp("MixMonitor");
-
- if (!monitor_options)
- monitor_options = "";
if (mixmonapp) {
+ ast_debug(1, "Starting MixMonitor as requested.\n");
+ if (!monitorfilename) {
+ if (qe->chan->cdr)
+ ast_copy_string(tmpid, qe->chan->cdr->uniqueid, sizeof(tmpid));
+ else
+ snprintf(tmpid, sizeof(tmpid), "chan-%lx", ast_random());
+ } else {
+ const char *m = monitorfilename;
+ for (p = tmpid2; p < tmpid2 + sizeof(tmpid2) - 1; p++, m++) {
+ switch (*m) {
+ case '^':
+ if (*(m + 1) == '{')
+ *p = '$';
+ break;
+ case ',':
+ *p++ = '\\';
+ /* Fall through */
+ default:
+ *p = *m;
+ }
+ if (*m == '\0')
+ break;
+ }
+ if (p == tmpid2 + sizeof(tmpid2))
+ tmpid2[sizeof(tmpid2) - 1] = '\0';
+
+ pbx_substitute_variables_helper(qe->chan, tmpid2, tmpid, sizeof(tmpid) - 1);
+ }
+
+ ast_channel_lock(qe->chan);
+ if ((monitor_exec = pbx_builtin_getvar_helper(qe->chan, "MONITOR_EXEC"))) {
+ monitor_exec = ast_strdupa(monitor_exec);
+ }
+ if ((monitor_options = pbx_builtin_getvar_helper(qe->chan, "MONITOR_OPTIONS"))) {
+ monitor_options = ast_strdupa(monitor_options);
+ } else {
+ monitor_options = "";
+ }
+ ast_channel_unlock(qe->chan);
+
+ if (monitor_exec) {
+ const char *m = monitor_exec;
+ for (p = meid2; p < meid2 + sizeof(meid2) - 1; p++, m++) {
+ switch (*m) {
+ case '^':
+ if (*(m + 1) == '{')
+ *p = '$';
+ break;
+ case ',':
+ *p++ = '\\';
+ /* Fall through */
+ default:
+ *p = *m;
+ }
+ if (*m == '\0')
+ break;
+ }
+ if (p == meid2 + sizeof(meid2))
+ meid2[sizeof(meid2) - 1] = '\0';
+
+ pbx_substitute_variables_helper(qe->chan, meid2, meid, sizeof(meid) - 1);
+ }
+
+ snprintf(tmpid2, sizeof(tmpid2), "%s.%s", tmpid, qe->parent->monfmt);
+
if (!ast_strlen_zero(monitor_exec))
snprintf(mixmonargs, sizeof(mixmonargs), "%s,b%s,%s", tmpid2, monitor_options, monitor_exec);
else
snprintf(mixmonargs, sizeof(mixmonargs), "%s,b%s", tmpid2, monitor_options);
-
+
ast_debug(1, "Arguments being passed to MixMonitor: %s\n", mixmonargs);
/* We purposely lock the CDR so that pbx_exec does not update the application data */
if (qe->chan->cdr)
@@ -3534,9 +3542,9 @@
if (qe->chan->cdr)
ast_clear_flag(qe->chan->cdr, AST_CDR_FLAG_LOCKED);
- } else
+ } else {
ast_log(LOG_WARNING, "Asked to run MixMonitor on this call, but cannot find the MixMonitor app!\n");
-
+ }
}
}
/* Drop out of the queue at this point, to prepare for next caller */
@@ -4449,6 +4457,7 @@
qe.expire = 0;
/* Get the priority from the variable ${QUEUE_PRIO} */
+ ast_channel_lock(chan);
user_priority = pbx_builtin_getvar_helper(chan, "QUEUE_PRIO");
if (user_priority) {
if (sscanf(user_priority, "%d", &prio) == 1) {
@@ -4488,6 +4497,7 @@
} else {
min_penalty = 0;
}
+ ast_channel_unlock(chan);
if (args.options && (strchr(args.options, 'r')))
ringing = 1;
Modified: team/jpeeler/chan_dahdi/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/apps/app_voicemail.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/apps/app_voicemail.c (original)
+++ team/jpeeler/chan_dahdi/apps/app_voicemail.c Thu May 29 14:34:13 2008
@@ -522,6 +522,7 @@
" the greeting. Context defaults to the current context.\n"
" g(#) - Use the specified amount of gain when recording the voicemail\n"
" message. The units are whole-number decibels (dB).\n"
+ " Only works on supported technologies, which is DAHDI only.\n"
" s - Skip the playback of instructions for leaving a message to the\n"
" calling party.\n"
" u - Play the 'unavailable' greeting.\n"
Modified: team/jpeeler/chan_dahdi/channels/chan_agent.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/channels/chan_agent.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/channels/chan_agent.c (original)
+++ team/jpeeler/chan_dahdi/channels/chan_agent.c Thu May 29 14:34:13 2008
@@ -1520,9 +1520,7 @@
ast_mutex_lock(&p->lock);
while (p->owner && ast_channel_trylock(p->owner)) {
- ast_mutex_unlock(&p->lock);
- usleep(1);
- ast_mutex_lock(&p->lock);
+ DEADLOCK_AVOIDANCE(&p->lock);
}
if (p->owner) {
ast_softhangup(p->owner, AST_SOFTHANGUP_EXPLICIT);
@@ -1530,9 +1528,7 @@
}
while (p->chan && ast_channel_trylock(p->chan)) {
- ast_mutex_unlock(&p->lock);
- usleep(1);
- ast_mutex_lock(&p->lock);
+ DEADLOCK_AVOIDANCE(&p->lock);
}
if (p->chan) {
ast_softhangup(p->chan, AST_SOFTHANGUP_EXPLICIT);
Modified: team/jpeeler/chan_dahdi/channels/chan_alsa.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/channels/chan_alsa.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/channels/chan_alsa.c (original)
+++ team/jpeeler/chan_dahdi/channels/chan_alsa.c Thu May 29 14:34:13 2008
@@ -297,9 +297,7 @@
static void grab_owner(void)
{
while (alsa.owner && ast_channel_trylock(alsa.owner)) {
- ast_mutex_unlock(&alsalock);
- usleep(1);
- ast_mutex_lock(&alsalock);
+ DEADLOCK_AVOIDANCE(&alsalock);
}
}
Modified: team/jpeeler/chan_dahdi/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/channels/chan_iax2.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/channels/chan_iax2.c (original)
+++ team/jpeeler/chan_dahdi/channels/chan_iax2.c Thu May 29 14:34:13 2008
@@ -1474,13 +1474,13 @@
#define NEW_ALLOW 1
#define NEW_FORCE 2
-static int match(struct sockaddr_in *sin, unsigned short callno, unsigned short dcallno, const struct chan_iax2_pvt *cur, int full_frame)
+static int match(struct sockaddr_in *sin, unsigned short callno, unsigned short dcallno, const struct chan_iax2_pvt *cur, int check_dcallno)
{
if ((cur->addr.sin_addr.s_addr == sin->sin_addr.s_addr) &&
(cur->addr.sin_port == sin->sin_port)) {
/* This is the main host */
if ( (cur->peercallno == 0 || cur->peercallno == callno) &&
- (full_frame ? dcallno == cur->callno : 1) ) {
+ (check_dcallno ? dcallno == cur->callno : 1) ) {
/* That's us. Be sure we keep track of the peer call number */
return 1;
}
@@ -1592,7 +1592,7 @@
/*
* \note Calling this function while holding another pvt lock can cause a deadlock.
*/
-static int __find_callno(unsigned short callno, unsigned short dcallno, struct sockaddr_in *sin, int new, int sockfd, int return_locked, int full_frame)
+static int __find_callno(unsigned short callno, unsigned short dcallno, struct sockaddr_in *sin, int new, int sockfd, int return_locked, int check_dcallno)
{
int res = 0;
int x;
@@ -1606,7 +1606,7 @@
.callno = dcallno,
.peercallno = callno,
/* hack!! */
- .frames_received = full_frame,
+ .frames_received = check_dcallno,
};
memcpy(&tmp_pvt.addr, sin, sizeof(tmp_pvt.addr));
@@ -1626,7 +1626,7 @@
ast_mutex_lock(&iaxsl[x]);
if (iaxs[x]) {
/* Look for an exact match */
- if (match(sin, callno, dcallno, iaxs[x], full_frame)) {
+ if (match(sin, callno, dcallno, iaxs[x], check_dcallno)) {
res = x;
}
}
@@ -1637,7 +1637,7 @@
ast_mutex_lock(&iaxsl[x]);
if (iaxs[x]) {
/* Look for an exact match */
- if (match(sin, callno, dcallno, iaxs[x], full_frame)) {
+ if (match(sin, callno, dcallno, iaxs[x], check_dcallno)) {
res = x;
}
}
@@ -1746,9 +1746,7 @@
if (iaxs[callno] && iaxs[callno]->owner) {
if (ast_channel_trylock(iaxs[callno]->owner)) {
/* Avoid deadlock by pausing and trying again */
- ast_mutex_unlock(&iaxsl[callno]);
- usleep(1);
- ast_mutex_lock(&iaxsl[callno]);
+ DEADLOCK_AVOIDANCE(&iaxsl[callno]);
} else {
ast_queue_frame(iaxs[callno]->owner, f);
ast_channel_unlock(iaxs[callno]->owner);
@@ -1779,9 +1777,7 @@
if (iaxs[callno] && iaxs[callno]->owner) {
if (ast_channel_trylock(iaxs[callno]->owner)) {
/* Avoid deadlock by pausing and trying again */
- ast_mutex_unlock(&iaxsl[callno]);
- usleep(1);
- ast_mutex_lock(&iaxsl[callno]);
+ DEADLOCK_AVOIDANCE(&iaxsl[callno]);
} else {
ast_queue_hangup(iaxs[callno]->owner);
ast_channel_unlock(iaxs[callno]->owner);
@@ -1813,9 +1809,7 @@
if (iaxs[callno] && iaxs[callno]->owner) {
if (ast_channel_trylock(iaxs[callno]->owner)) {
/* Avoid deadlock by pausing and trying again */
- ast_mutex_unlock(&iaxsl[callno]);
- usleep(1);
- ast_mutex_lock(&iaxsl[callno]);
+ DEADLOCK_AVOIDANCE(&iaxsl[callno]);
} else {
ast_queue_control_data(iaxs[callno]->owner, control, data, datalen);
ast_channel_unlock(iaxs[callno]->owner);
@@ -3773,9 +3767,7 @@
{
ast_mutex_lock(&iaxsl[callno0]);
while (ast_mutex_trylock(&iaxsl[callno1])) {
- ast_mutex_unlock(&iaxsl[callno0]);
- usleep(10);
- ast_mutex_lock(&iaxsl[callno0]);
+ DEADLOCK_AVOIDANCE(&iaxsl[callno0]);
}
}
@@ -3969,9 +3961,7 @@
/* We don't know the remote side's call number, yet. :( */
int count = 10;
while (count-- && pvt && !pvt->peercallno) {
- ast_mutex_unlock(&iaxsl[callno]);
- usleep(1);
- ast_mutex_lock(&iaxsl[callno]);
+ DEADLOCK_AVOIDANCE(&iaxsl[callno]);
pvt = iaxs[callno];
}
if (!pvt->peercallno) {
@@ -8008,8 +7998,10 @@
f.subclass = 0;
}
- if (!fr->callno)
- fr->callno = find_callno(ntohs(mh->callno) & ~IAX_FLAG_FULL, dcallno, &sin, new, fd, ntohs(mh->callno) & IAX_FLAG_FULL);
+ if (!fr->callno) {
+ fr->callno = find_callno(ntohs(mh->callno) & ~IAX_FLAG_FULL, dcallno, &sin, new, fd,
+ (ntohs(mh->callno) & IAX_FLAG_FULL) && f.frametype == AST_FRAME_IAX && f.subclass == IAX_COMMAND_ACK);
+ }
if (fr->callno > 0)
ast_mutex_lock(&iaxsl[fr->callno]);
@@ -8156,12 +8148,11 @@
(f.frametype != AST_FRAME_IAX))) {
unsigned char x;
int call_to_destroy;
- /* XXX This code is not very efficient. Surely there is a better way which still
- properly handles boundary conditions? XXX */
/* First we have to qualify that the ACKed value is within our window */
- for (x=iaxs[fr->callno]->rseqno; x != iaxs[fr->callno]->oseqno; x++)
- if (fr->iseqno == x)
- break;
+ if (iaxs[fr->callno]->rseqno >= iaxs[fr->callno]->oseqno || (fr->iseqno >= iaxs[fr->callno]->rseqno && fr->iseqno < iaxs[fr->callno]->oseqno))
+ x = fr->iseqno;
+ else
+ x = iaxs[fr->callno]->oseqno;
if ((x != iaxs[fr->callno]->oseqno) || (iaxs[fr->callno]->oseqno == fr->iseqno)) {
/* The acknowledgement is within our window. Time to acknowledge everything
that it says to */
@@ -8285,9 +8276,7 @@
int orignative;
retryowner:
if (ast_channel_trylock(iaxs[fr->callno]->owner)) {
- ast_mutex_unlock(&iaxsl[fr->callno]);
- usleep(1);
- ast_mutex_lock(&iaxsl[fr->callno]);
+ DEADLOCK_AVOIDANCE(&iaxsl[fr->callno]);
if (iaxs[fr->callno] && iaxs[fr->callno]->owner) goto retryowner;
}
if (iaxs[fr->callno]) {
@@ -8729,9 +8718,7 @@
ast_verb(3, "Format for call is %s\n", ast_getformatname(iaxs[fr->callno]->owner->nativeformats));
retryowner2:
if (ast_channel_trylock(iaxs[fr->callno]->owner)) {
- ast_mutex_unlock(&iaxsl[fr->callno]);
- usleep(1);
- ast_mutex_lock(&iaxsl[fr->callno]);
+ DEADLOCK_AVOIDANCE(&iaxsl[fr->callno]);
if (iaxs[fr->callno] && iaxs[fr->callno]->owner) goto retryowner2;
}
@@ -9909,7 +9896,11 @@
peer_unref(peer);
/* And send the poke */
- send_command(iaxs[peer->callno], AST_FRAME_IAX, IAX_COMMAND_POKE, 0, NULL, 0, -1);
+ ast_mutex_lock(&iaxsl[peer->callno]);
+ if (iaxs[peer->callno]) {
+ send_command(iaxs[peer->callno], AST_FRAME_IAX, IAX_COMMAND_POKE, 0, NULL, 0, -1);
+ }
+ ast_mutex_unlock(&iaxsl[peer->callno]);
return 0;
}
Modified: team/jpeeler/chan_dahdi/channels/chan_jingle.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/channels/chan_jingle.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/channels/chan_jingle.c (original)
+++ team/jpeeler/chan_dahdi/channels/chan_jingle.c Thu May 29 14:34:13 2008
@@ -111,7 +111,9 @@
char cid_name[80]; /*!< Caller ID name */
char exten[80]; /*!< Called extension */
struct ast_channel *owner; /*!< Master Channel */
+ char audio_content_name[100]; /*!< name attribute of content tag */
struct ast_rtp *rtp; /*!< RTP audio session */
+ char video_content_name[100]; /*!< name attribute of content tag */
struct ast_rtp *vrtp; /*!< RTP video session */
int jointcapability; /*!< Supported capability at both ends (codecs ) */
int peercapability;
@@ -956,7 +958,7 @@
struct jingle_pvt *p, *tmp = client->p;
struct ast_channel *chan;
int res;
- iks *payload_type;
+ iks *codec, *content, *description;
/* Make sure our new call doesn't exist yet */
while (tmp) {
@@ -974,45 +976,74 @@
return -1;
}
chan = jingle_new(client, p, AST_STATE_DOWN, pak->from->user);
- if (chan) {
- ast_mutex_lock(&p->lock);
- ast_copy_string(p->them, pak->from->full, sizeof(p->them));
- if (iks_find_attrib(pak->query, JINGLE_SID)) {
- ast_copy_string(p->sid, iks_find_attrib(pak->query, JINGLE_SID),
- sizeof(p->sid));
- }
-
- payload_type = iks_child(iks_child(iks_child(iks_child(pak->x))));
- while (payload_type) {
- ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(payload_type, "id")));
- ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(payload_type, "id")), "audio", iks_find_attrib(payload_type, "name"), 0);
- payload_type = iks_next(payload_type);
- }
-
- ast_mutex_unlock(&p->lock);
- ast_setstate(chan, AST_STATE_RING);
- res = ast_pbx_start(chan);
-
- switch (res) {
- case AST_PBX_FAILED:
- ast_log(LOG_WARNING, "Failed to start PBX :(\n");
- jingle_response(client, pak, "service-unavailable", NULL);
- break;
- case AST_PBX_CALL_LIMIT:
- ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
- jingle_response(client, pak, "service-unavailable", NULL);
- break;
- case AST_PBX_SUCCESS:
- jingle_response(client, pak, NULL, NULL);
- jingle_create_candidates(client, p,
- iks_find_attrib(pak->query, JINGLE_SID),
- iks_find_attrib(pak->x, "from"));
- /* nothing to do */
- break;
- }
- } else {
+ if (!chan) {
jingle_free_pvt(client, p);
- }
+ return -1;
+ }
+ ast_mutex_lock(&p->lock);
+ ast_copy_string(p->them, pak->from->full, sizeof(p->them));
+ if (iks_find_attrib(pak->query, JINGLE_SID)) {
+ ast_copy_string(p->sid, iks_find_attrib(pak->query, JINGLE_SID),
+ sizeof(p->sid));
+ }
+
+ /* content points to the first <content/> tag */
+ content = iks_child(iks_child(pak->x));
+ while (content) {
+ description = iks_find_with_attrib(content, "description", "xmlns", JINGLE_AUDIO_RTP_NS);
+ if (description) {
+ /* audio content found */
+ codec = iks_child(iks_child(content));
+ ast_copy_string(p->audio_content_name, iks_find_attrib(content, "name"), sizeof(p->audio_content_name));
+
+ while (codec) {
+ ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ codec = iks_next(codec);
+ }
+ }
+
+ description = NULL;
+ codec = NULL;
+
+ description = iks_find_with_attrib(content, "description", "xmlns", JINGLE_VIDEO_RTP_NS);
+ if (description) {
+ /* video content found */
+ codec = iks_child(iks_child(content));
+ ast_copy_string(p->video_content_name, iks_find_attrib(content, "name"), sizeof(p->video_content_name));
+
+ while (codec) {
+ ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ codec = iks_next(codec);
+ }
+ }
+
+ content = iks_next(content);
+ }
+
+ ast_mutex_unlock(&p->lock);
+ ast_setstate(chan, AST_STATE_RING);
+ res = ast_pbx_start(chan);
+
+ switch (res) {
+ case AST_PBX_FAILED:
+ ast_log(LOG_WARNING, "Failed to start PBX :(\n");
+ jingle_response(client, pak, "service-unavailable", NULL);
+ break;
+ case AST_PBX_CALL_LIMIT:
+ ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
+ jingle_response(client, pak, "service-unavailable", NULL);
+ break;
+ case AST_PBX_SUCCESS:
+ jingle_response(client, pak, NULL, NULL);
+ jingle_create_candidates(client, p,
+ iks_find_attrib(pak->query, JINGLE_SID),
+ iks_find_attrib(pak->x, "from"));
+ /* nothing to do */
+ break;
+ }
+
return 1;
}
@@ -1316,6 +1347,8 @@
payload_pcmu = iks_new("payload-type");
payload_eg711u = iks_new("payload-type");
+ ast_copy_string(p->audio_content_name, "asterisk-audio-content", sizeof(p->audio_content_name));
+
iks_insert_attrib(iq, "type", "set");
iks_insert_attrib(iq, "to", p->them);
iks_insert_attrib(iq, "from", client->jid->full);
@@ -1325,8 +1358,10 @@
iks_insert_attrib(jingle, JINGLE_SID, p->sid);
iks_insert_attrib(jingle, "initiator", client->jid->full);
iks_insert_attrib(jingle, "xmlns", JINGLE_NS);
+
+ /* For now, we only send one audio based content */
iks_insert_attrib(content, "creator", "initiator");
- iks_insert_attrib(content, "name", "asterisk-audio-content");
+ iks_insert_attrib(content, "name", p->audio_content_name);
iks_insert_attrib(content, "profile", "RTP/AVP");
iks_insert_attrib(description, "xmlns", JINGLE_AUDIO_RTP_NS);
iks_insert_attrib(transport, "xmlns", JINGLE_ICE_UDP_NS);
@@ -1334,7 +1369,6 @@
iks_insert_attrib(payload_pcmu, "name", "PCMU");
iks_insert_attrib(payload_eg711u, "id", "100");
iks_insert_attrib(payload_eg711u, "name", "EG711U");
-
iks_insert_node(description, payload_pcmu);
iks_insert_node(description, payload_eg711u);
iks_insert_node(content, description);
Modified: team/jpeeler/chan_dahdi/channels/chan_mgcp.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/channels/chan_mgcp.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/channels/chan_mgcp.c (original)
+++ team/jpeeler/chan_dahdi/channels/chan_mgcp.c Thu May 29 14:34:13 2008
@@ -596,9 +596,7 @@
ast_channel_unlock(sub->owner);
break;
} else {
- ast_mutex_unlock(&sub->lock);
- usleep(1);
- ast_mutex_lock(&sub->lock);
+ DEADLOCK_AVOIDANCE(&sub->lock);
}
} else
break;
@@ -614,9 +612,7 @@
ast_channel_unlock(sub->owner);
break;
} else {
- ast_mutex_unlock(&sub->lock);
- usleep(1);
- ast_mutex_lock(&sub->lock);
+ DEADLOCK_AVOIDANCE(&sub->lock);
}
} else
break;
Modified: team/jpeeler/chan_dahdi/channels/chan_phone.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/channels/chan_phone.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/channels/chan_phone.c (original)
+++ team/jpeeler/chan_dahdi/channels/chan_phone.c Thu May 29 14:34:13 2008
@@ -59,7 +59,7 @@
#include "asterisk/stringfields.h"
#include "asterisk/musiconhold.h"
-#include "DialTone.h"
+#include "chan_phone.h"
#ifdef QTI_PHONEJACK_TJ_PCI /* check for the newer quicknet driver v.3.1.0 which has this symbol */
#define QNDRV_VER 310
Modified: team/jpeeler/chan_dahdi/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/channels/chan_sip.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/channels/chan_sip.c (original)
+++ team/jpeeler/chan_dahdi/channels/chan_sip.c Thu May 29 14:34:13 2008
@@ -1010,11 +1010,12 @@
#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< DP: Compensate for buggy RFC2833 implementations */
#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< DP: Buggy CISCO MWI fix */
#define SIP_PAGE2_REGISTERTRYING (1 << 29) /*!< DP: Send 100 Trying on REGISTER attempts */
+#define SIP_PAGE2_UDPTL_DESTINATION (1 << 30) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
- SIP_PAGE2_TEXTSUPPORT )
+ SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION)
/*@}*/
@@ -6078,6 +6079,7 @@
ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
p->maxcallbitrate = default_maxcallbitrate;
p->autoframing = global_autoframing;
+ ast_rtp_codec_setpref(p->rtp, &p->prefs);
}
if (useglobal_nat && sin) {
@@ -6888,6 +6890,16 @@
if (p->udptl) {
if (udptlportno > 0) {
sin.sin_port = htons(udptlportno);
+ if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
+ struct sockaddr_in peer;
+ ast_rtp_get_peer(p->rtp, &peer);
+ if (peer.sin_addr.s_addr) {
+ memcpy(&sin.sin_addr, &peer.sin_addr, sizeof(&sin.sin_addr));
+ if (debug) {
+ ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(sin.sin_addr));
+ }
+ }
+ }
ast_udptl_set_peer(p->udptl, &sin);
if (debug)
ast_debug(1, "Peer T.38 UDPTL is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
@@ -6989,23 +7001,20 @@
ast_debug(1, "Can't read framing from SDP: %s\n", a);
}
}
- if (framing && last_rtpmap_codec) {
- if (p->autoframing) {
- struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
- int codec_n;
- int format = 0;
- for (codec_n = 0; codec_n < last_rtpmap_codec; codec_n++) {
- format = ast_rtp_codec_getformat(found_rtpmap_codecs[codec_n]);
- if (!format) /* non-codec or not found */
- continue;
- ast_debug(1, "Setting framing for %d to %ld\n", format, framing);
- ast_codec_pref_setsize(pref, format, framing);
- }
- ast_rtp_codec_setpref(p->rtp, pref);
+ if (framing && p->autoframing) {
+ struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
+ int codec_n;
+ int format = 0;
+ for (codec_n = 0; codec_n < MAX_RTP_PT; codec_n++) {
+ format = ast_rtp_codec_getformat(codec_n);
+ if (!format) /* non-codec or not found */
+ continue;
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing);
+ ast_codec_pref_setsize(pref, format, framing);
}
+ ast_rtp_codec_setpref(p->rtp, pref);
}
- memset(&found_rtpmap_codecs, 0, sizeof(found_rtpmap_codecs));
- last_rtpmap_codec = 0;
continue;
} else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
@@ -20457,6 +20466,9 @@
} else if (!strcasecmp(v->name, "buggymwi")) {
ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
+ } else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
+ ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
+ ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
} else
res = 0;
Modified: team/jpeeler/chan_dahdi/channels/chan_skinny.c
URL: http://svn.digium.com/view/asterisk/team/jpeeler/chan_dahdi/channels/chan_skinny.c?view=diff&rev=119069&r1=119068&r2=119069
==============================================================================
--- team/jpeeler/chan_dahdi/channels/chan_skinny.c (original)
+++ team/jpeeler/chan_dahdi/channels/chan_skinny.c Thu May 29 14:34:13 2008
@@ -63,7 +63,6 @@
#include "asterisk/utils.h"
#include "asterisk/dsp.h"
#include "asterisk/stringfields.h"
-#include "asterisk/astobj.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/threadstorage.h"
#include "asterisk/devicestate.h"
@@ -88,9 +87,9 @@
SKINNY_CODEC_H263 = 101
};
-#define DEFAULT_SKINNY_PORT 2000
-#define DEFAULT_SKINNY_BACKLOG 2
-#define SKINNY_MAX_PACKET 1000
+#define DEFAULT_SKINNY_PORT 2000
+#define DEFAULT_SKINNY_BACKLOG 2
+#define SKINNY_MAX_PACKET 1000
static unsigned int tos = 0;
static unsigned int tos_audio = 0;
@@ -100,9 +99,9 @@
static unsigned int cos_video = 0;
static int keep_alive = 120;
-static char vmexten[AST_MAX_EXTENSION]; /* Voicemail pilot number */
-static char used_context[AST_MAX_EXTENSION]; /* Voicemail pilot number */
-static char regcontext[AST_MAX_CONTEXT]; /* Context for auto-extension */
+static char vmexten[AST_MAX_EXTENSION]; /* Voicemail pilot number */
[... 1113 lines stripped ...]
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