if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
- ast_log(LOG_WARNING, "dahdi_call called on %s, neither down nor reserved\n", ast->name);
+ ast_log(LOG_WARNING, "analog_call called on %s, neither down nor reserved\n", ast->name);
return -1;
}
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Posted: Tue May 19, 2009 7:59 pm Post subject: [svn-commits] jpeeler: branch jpeeler/asterisk-sigwork-trunk
Author: jpeeler
Date: Tue May 19 15:52:58 2009
New Revision: 195588
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=195588
Log:
Free and set to null cidspill when using sig_analog. Also, add locking to do_monitor such that when building the poll fd array to not erroneously set POLLIN on a channel that is being hung up.
if ((chan = dahdi_new(mtd->pvt, AST_STATE_RING, 0, SUB_REAL, 0, 0))) {
- if (ast_pthread_create_detached(&threadid, NULL, analog_ss_thread, chan)) {
+ if (analog_ss_thread_start(mtd->pvt->sig_pvt, chan)) {
ast_log(LOG_WARNING, "Unable to start simple switch thread on channel %d\n", mtd->pvt->channel);
res = tone_zone_play_tone(mtd->pvt->subs[SUB_REAL].dfd, DAHDI_TONE_CONGESTION);
if (res < 0)
@@ -10554,7 +10553,7 @@
if (!p)
ast_log(LOG_ERROR, "No sig_pvt?\n");
- if (!p->owner && !p->subs[ANALOG_SUB_REAL].owner) {
+ if (!p->owner && !p->subs[SUB_REAL].owner) {
/* This needs to be watched, as it lacks an owner */
pfds[count].fd = i->subs[SUB_REAL].dfd;
pfds[count].events = POLLPRI;
@@ -10564,7 +10563,7 @@
pfds[count].events |= POLLIN;
count++;
}
- } else {
+ } else {
if (!i->owner && !i->subs[SUB_REAL].owner && !i->mwimonitoractive ) {
/* This needs to be watched, as it lacks an owner */
pfds[count].fd = i->subs[SUB_REAL].dfd;
Posted: Fri May 22, 2009 3:57 pm Post subject: [svn-commits] jpeeler: branch jpeeler/asterisk-sigwork-trunk
Author: jpeeler
Date: Fri May 22 11:54:11 2009
New Revision: 196273
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=196273
Log:
Add callbacks to increase/decrease analog_ss_thread counts from within sig_analog. This is a lot of code to do so little, but should be done this way until the analog_ss_thread in chan_dahdi completely goes away.
struct dahdi_pvt *round_robin[32];
@@ -15736,8 +15753,8 @@
if (p->owner)
ioctl(p->subs[SUB_REAL].dfd, DAHDI_HOOK, &x); /* important to create an event for dahdi_wait_event to register so that all analog_ss_threads terminate */
}
- ast_cond_wait(&ss_thread_complete, &ss_thread_lock);
- }
+ ast_cond_wait(&ss_thread_complete, &ss_thread_lock);
+ }
/* ensure any created channels before monitor threads were stopped are hungup */
dahdi_softhangup_all();
Modified: team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.c?view=diff&rev=196273&r1=196272&r2=196273
==============================================================================
--- team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.c (original)
+++ team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.c Fri May 22 11:54:11 2009
@@ -70,7 +70,6 @@
/* "smdi" is intentionally not supported here, as there is a much better
* way to do this in the dialplan now. */
};
-
/* in the bizarre case where the channel has become a zombie before we
@@ -1258,7 +1277,7 @@
if (!p) {
ast_log(LOG_WARNING, "Channel became a zombie before simple switch could be started (%s)\n", chan->name);
ast_hangup(chan);
- return NULL;
+ goto quit;
}
if (option_verbose > 2)
@@ -1267,7 +1286,7 @@
if (index < 0) {
ast_log(LOG_WARNING, "Huh?\n");
ast_hangup(chan);
- return NULL;
+ goto quit;
}
analog_dsp_reset_and_flush_digits(p);
switch (p->sig) {
@@ -1283,7 +1302,7 @@
case ANALOG_SIG_SF_FEATB:
case ANALOG_SIG_SFWINK:
if (analog_wink(p, index))
- return NULL;
+ goto quit;
/* Fall through */
case ANALOG_SIG_EM:
case ANALOG_SIG_EM_E1:
@@ -1321,7 +1340,7 @@
res = analog_my_getsigstr(chan, dtmfbuf + 1, "#", 3000);
if (res < 1)
analog_dsp_reset_and_flush_digits(p);
- if (analog_wink(p, index)) return NULL;
+ if (analog_wink(p, index)) goto quit;
dtmfbuf[0] = 0;
/* Wait for the first digit (up to 5 seconds). */
res = ast_waitfordigit(chan, 5000);
@@ -1336,7 +1355,7 @@
/* if international caca, do it again to get real ANO */
if ((p->sig == ANALOG_SIG_FEATDMF) && (dtmfbuf[1] != '0') && (strlen(dtmfbuf) != 14))
{
- if (analog_wink(p, index)) return NULL;
+ if (analog_wink(p, index)) goto quit;
dtmfbuf[0] = 0;
/* Wait for the first digit (up to 5 seconds). */
res = ast_waitfordigit(chan, 5000);
@@ -1386,7 +1405,7 @@
if (res < 0) {
ast_debug(1, "waitfordigit returned < 0...\n");
ast_hangup(chan);
- return NULL;
+ goto quit;
} else if (res) {
dtmfbuf[len++] = res;
dtmfbuf[len] = '\0';
@@ -1400,11 +1419,11 @@
if (res == -1) {
ast_log(LOG_WARNING, "getdtmf on channel %d: %s\n", p->channel, strerror(errno));
ast_hangup(chan);
- return NULL;
+ goto quit;
} else if (res < 0) {
ast_debug(1, "Got hung up before digits finished\n");
ast_hangup(chan);
- return NULL;
+ goto quit;
}
if (p->sig == ANALOG_SIG_FGC_CAMA) {
@@ -1412,7 +1431,7 @@
if (ast_safe_sleep(chan,1000) == -1) {
ast_hangup(chan);
- return NULL;
+ goto quit;
}
analog_off_hook(p);
analog_dsp_set_digitmode(p, ANALOG_DIGITMODE_MF);
@@ -1501,7 +1520,7 @@
/* some switches require a minimum guard time between
the last FGD wink and something that answers
immediately. This ensures it */
- if (ast_safe_sleep(chan,100)) return NULL;
+ if (ast_safe_sleep(chan,100)) goto quit;
}
analog_set_echocanceller(p, 1);
@@ -1515,7 +1534,7 @@
ast_log(LOG_WARNING, "PBX exited non-zero\n");
res = analog_play_tone(p, index, ANALOG_TONE_CONGESTION);
}
- return NULL;
+ goto quit;
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_2 "Unknown extension '%s' in context '%s' requested\n", exten, chan->context);
@@ -1530,7 +1549,7 @@
ast_waitstream(chan, "");
res = analog_play_tone(p, index, ANALOG_TONE_CONGESTION);
ast_hangup(chan);
- return NULL;
+ goto quit;
}
break;
case ANALOG_SIG_FXOLS:
@@ -1554,7 +1573,7 @@
ast_debug(1, "waitfordigit returned < 0...\n");
res = analog_play_tone(p, index, -1);
ast_hangup(chan);
- return NULL;
+ goto quit;
} else if (res) {
exten[len++]=res;
exten[len] = '\0';
@@ -1600,7 +1619,7 @@
ast_log(LOG_WARNING, "PBX exited non-zero\n");
res = analog_play_tone(p, index, ANALOG_TONE_CONGESTION);
}
- return NULL;
+ goto quit;
}
} else {
/* It's a match, but they just typed a digit, and there is an ambiguous match,
@@ -1612,7 +1631,7 @@
res = analog_play_tone(p, index, ANALOG_TONE_CONGESTION);
analog_wait_event(p);
ast_hangup(chan);
- return NULL;
+ goto quit;
} else if (p->callwaiting && !strcmp(exten, "*70")) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Disabling call waiting on %s\n", chan->name);
@@ -1648,11 +1667,11 @@
analog_wait_event(p);
}
ast_hangup(chan);
- return NULL;
+ goto quit;
} else {
ast_log(LOG_WARNING, "Huh? Got *8# on call not on real\n");
ast_hangup(chan);
- return NULL;
+ goto quit;
}
} else if (!p->hidecallerid && !strcmp(exten, "*67")) {
@@ -1787,7 +1806,7 @@
if (ast_bridged_channel(p->subs[ANALOG_SUB_REAL].owner))
ast_queue_control(p->subs[ANALOG_SUB_REAL].owner, AST_CONTROL_UNHOLD);
ast_hangup(chan);
- return NULL;
+ goto quit;
} else {
analog_play_tone(p, index, ANALOG_TONE_CONGESTION);
analog_wait_event(p);
@@ -1796,7 +1815,7 @@
analog_unalloc_sub(p, ANALOG_SUB_THREEWAY);
p->owner = p->subs[ANALOG_SUB_REAL].owner;
ast_hangup(chan);
- return NULL;
+ goto quit;
}
#endif
} else if (!ast_canmatch_extension(chan, chan->context, exten, 1, chan->cid.cid_num) &&
@@ -1835,7 +1854,7 @@
ast_log(LOG_WARNING, "DTMFCID timed out waiting for ring. "
"Exiting simple switch\n");
ast_hangup(chan);
- return NULL;
+ goto quit;
}
f = ast_read(chan);
if (!f)
@@ -1883,7 +1902,7 @@
ast_log(LOG_WARNING, "I/O MUX failed: %s\n", strerror(errno));
callerid_free(cs);
ast_hangup(chan);
- return NULL;
+ goto quit;
}
if (i & DAHDI_IOMUX_SIGEVENT) {
res = dahdi_get_event(p->subs[index].dfd);
@@ -1907,7 +1926,7 @@
ast_log(LOG_WARNING, "read returned error: %s\n", strerror(errno));
callerid_free(cs);
ast_hangup(chan);
- return NULL;
+ goto quit;
}
break;
}
@@ -1950,12 +1969,12 @@
ast_log(LOG_WARNING, "CID timed out waiting for ring. "
"Exiting simple switch\n");
ast_hangup(chan);
- return NULL;
+ goto quit;
}
if (!(f = ast_read(chan))) {
ast_log(LOG_WARNING, "Hangup received waiting for ring. Exiting simple switch\n");
ast_hangup(chan);
- return NULL;
+ goto quit;
}
ast_frfree(f);
if (chan->_state == AST_STATE_RING ||
@@ -1978,7 +1997,7 @@
"restarted by the actual ring.\n",
chan->name);
ast_hangup(chan);
- return NULL;
+ goto quit;
}
} else if (p->use_callerid && p->cid_start == ANALOG_CID_START_RING) {
int timeout = 10000; /* Ten seconds */
@@ -2001,7 +2020,7 @@
ast_debug(1, "Hanging up due to polarity reversal on channel %d while detecting callerid\n", p->channel);
p->polarity = POLARITY_IDLE;
ast_hangup(chan);
- return NULL;
+ goto quit;
} else if (ev != ANALOG_EVENT_NONE) {
break;
}
@@ -2049,7 +2068,7 @@
ast_log(LOG_WARNING, "I/O MUX failed: %s\n", strerror(errno));
callerid_free(cs);
ast_hangup(chan);
- return NULL;
+ goto quit;
}
if (i & DAHDI_IOMUX_SIGEVENT) {
res = dahdi_get_event(p->subs[index].dfd);
@@ -2060,7 +2079,7 @@
p->polarity = POLARITY_IDLE;
callerid_free(cs);
ast_hangup(chan);
- return NULL;
+ goto quit;
}
res = 0;
/* Let us detect callerid when the telco uses distinctive ring */
@@ -2080,7 +2099,7 @@
ast_log(LOG_WARNING, "read returned error: %s\n", strerror(errno));
callerid_free(cs);
ast_hangup(chan);
- return NULL;
+ goto quit;
}
break;
}
@@ -2141,7 +2160,7 @@
ast_hangup(chan);
ast_log(LOG_WARNING, "PBX exited non-zero\n");
}
- return NULL;
+ goto quit;
default:
ast_log(LOG_WARNING, "Don't know how to handle simple switch with signalling %s on channel %d\n", analog_sigtype_to_str(p->sig), p->channel);
res = analog_play_tone(p, index, ANALOG_TONE_CONGESTION);
@@ -2152,6 +2171,8 @@
if (res < 0)
ast_log(LOG_WARNING, "Unable to play congestion tone on channel %d\n", p->channel);
ast_hangup(chan);
+quit:
+ analog_decrease_ss_count(p);
return NULL;
}
Modified: team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.h
URL: http://svn.asterisk.org/svn-view/asterisk/team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.h?view=diff&rev=196273&r1=196272&r2=196273
==============================================================================
--- team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.h (original)
+++ team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.h Fri May 22 11:54:11 2009
@@ -165,6 +165,9 @@
int (* const check_for_conference)(void *pvt);
void (* const handle_notify_message)(struct ast_channel *chan, void *pvt, int cid_flags, int neon_mwievent);
+ /* callbacks for increasing and decreasing ss_thread_count, will handle locking and condition signal */
+ void (* const increase_ss_count)(void);
+ void (* const decrease_ss_count)(void);
};
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Modified: team/jpeeler/asterisk-sigwork-trunk/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/jpeeler/asterisk-sigwork-trunk/channels/chan_sip.c?view=diff&rev=196283&r1=196282&r2=196283
==============================================================================
--- team/jpeeler/asterisk-sigwork-trunk/channels/chan_sip.c (original)
+++ team/jpeeler/asterisk-sigwork-trunk/channels/chan_sip.c Fri May 22 12:18:59 2009
@@ -20350,7 +20350,7 @@
}
} else {
/* Notify the first other party that they are connected to someone else assuming that target.chan1
- has progressed far enough through the dialplan to have it's called party information set. */
+ has progressed far enough through the dialplan to have its called party information set. */
if (current->chan2) {
ast_channel_lock(target.chan1);
ast_party_connected_line_copy(&connected_caller, &target.chan1->connected);
Modified: team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.c?view=diff&rev=196619&r1=196618&r2=196619
==============================================================================
--- team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.c (original)
+++ team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.c Sun May 24 16:27:19 2009
@@ -1,9 +1,26 @@
/*
- * sig_analog.c -- Analog signalling module
+ * Asterisk -- An open source telephony toolkit.
*
- * Copyright (C) 2008, Digium, Inc.
+ * Copyright (C) 1999 - 2009, Digium, Inc.
*
- * Matthew Fredrickson <creslin@digium.com>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Analog signaling module
+ *
+ * \author Matthew Fredrickson <creslin@digium.com>
*/
#include "asterisk.h"
Modified: team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.h
URL: http://svn.asterisk.org/svn-view/asterisk/team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.h?view=diff&rev=196619&r1=196618&r2=196619
==============================================================================
--- team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.h (original)
+++ team/jpeeler/asterisk-sigwork-trunk/channels/sig_analog.h Sun May 24 16:27:19 2009
@@ -1,11 +1,28 @@
#ifndef _SIG_ANALOG_H
#define _SIG_ANALOG_H
/*
- * sig_analog.h -- Interface header for analog signalling module
- *
- * Copyright (C) 2008, Digium, Inc.
- *
- * Matthew Fredrickson <creslin@digium.com>
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2009, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Interface header for analog signaling module
+ *
+ * \author Matthew Fredrickson <creslin@digium.com>
*/
#include "asterisk/channel.h"
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Modified: team/jpeeler/asterisk-sigwork-trunk/channels/sig_pri.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/jpeeler/asterisk-sigwork-trunk/channels/sig_pri.c?view=diff&rev=196985&r1=196984&r2=196985
==============================================================================
--- team/jpeeler/asterisk-sigwork-trunk/channels/sig_pri.c (original)
+++ team/jpeeler/asterisk-sigwork-trunk/channels/sig_pri.c Tue May 26 18:29:53 2009
@@ -1,9 +1,26 @@
/*
- * sig_pri.c -- PRI signalling module
+ * Asterisk -- An open source telephony toolkit.
*
- * Copyright (C) 2008, Digium, Inc.
+ * Copyright (C) 1999 - 2009, Digium, Inc.
*
- * Matthew Fredrickson <creslin@digium.com>
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief PRI signaling module
+ *
+ * \author Matthew Fredrickson <creslin@digium.com>
*/
#include "asterisk.h"
Modified: team/jpeeler/asterisk-sigwork-trunk/channels/sig_pri.h
URL: http://svn.asterisk.org/svn-view/asterisk/team/jpeeler/asterisk-sigwork-trunk/channels/sig_pri.h?view=diff&rev=196985&r1=196984&r2=196985
==============================================================================
--- team/jpeeler/asterisk-sigwork-trunk/channels/sig_pri.h (original)
+++ team/jpeeler/asterisk-sigwork-trunk/channels/sig_pri.h Tue May 26 18:29:53 2009
@@ -1,11 +1,28 @@
#ifndef _SIG_PRI_H
#define _SIG_PRI_H
/*
- * sig_analog.h -- Interface header for analog signalling module
- *
- * Copyright (C) 2008, Digium, Inc.
- *
- * Matthew Fredrickson <creslin@digium.com>
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2009, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Interface header for PRI signaling module
+ *
+ * \author Matthew Fredrickson <creslin@digium.com>
*/
#include "asterisk/channel.h"
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
@@ -2133,48 +2132,48 @@
enum analog_event res = ANALOG_EVENT_ERROR;
switch (event) {
- case DAHDI_EVENT_DIALCOMPLETE:
- res = ANALOG_EVENT_DIALCOMPLETE;
- break;
- case DAHDI_EVENT_WINKFLASH:
- res = ANALOG_EVENT_WINKFLASH;
- break;
- case DAHDI_EVENT_ONHOOK:
- res = ANALOG_EVENT_ONHOOK;
- break;
- case DAHDI_EVENT_RINGOFFHOOK:
- res = ANALOG_EVENT_RINGOFFHOOK;
- break;
- case DAHDI_EVENT_ALARM:
- res = ANALOG_EVENT_ALARM;
- break;
- case DAHDI_EVENT_NOALARM:
- res = ANALOG_EVENT_NOALARM;
- break;
- case DAHDI_EVENT_HOOKCOMPLETE:
- res = ANALOG_EVENT_HOOKCOMPLETE;
- break;
- case DAHDI_EVENT_POLARITY:
- res = ANALOG_EVENT_POLARITY;
- break;
- case DAHDI_EVENT_RINGERON:
- res = ANALOG_EVENT_RINGERON;
- break;
- case DAHDI_EVENT_RINGEROFF:
- res = ANALOG_EVENT_RINGEROFF;
- break;
- case DAHDI_EVENT_RINGBEGIN:
- res = ANALOG_EVENT_RINGBEGIN;
- break;
- case DAHDI_EVENT_PULSE_START:
- res = ANALOG_EVENT_PULSE_START;
- break;
- case DAHDI_EVENT_NEONMWI_ACTIVE:
- res = ANALOG_EVENT_NEONMWI_ACTIVE;
- break;
- case DAHDI_EVENT_NEONMWI_INACTIVE:
- res = ANALOG_EVENT_NEONMWI_INACTIVE;
- break;
+ case DAHDI_EVENT_DIALCOMPLETE:
+ res = ANALOG_EVENT_DIALCOMPLETE;
+ break;
+ case DAHDI_EVENT_WINKFLASH:
+ res = ANALOG_EVENT_WINKFLASH;
+ break;
+ case DAHDI_EVENT_ONHOOK:
+ res = ANALOG_EVENT_ONHOOK;
+ break;
+ case DAHDI_EVENT_RINGOFFHOOK:
+ res = ANALOG_EVENT_RINGOFFHOOK;
+ break;
+ case DAHDI_EVENT_ALARM:
+ res = ANALOG_EVENT_ALARM;
+ break;
+ case DAHDI_EVENT_NOALARM:
+ res = ANALOG_EVENT_NOALARM;
+ break;
+ case DAHDI_EVENT_HOOKCOMPLETE:
+ res = ANALOG_EVENT_HOOKCOMPLETE;
+ break;
+ case DAHDI_EVENT_POLARITY:
+ res = ANALOG_EVENT_POLARITY;
+ break;
+ case DAHDI_EVENT_RINGERON:
+ res = ANALOG_EVENT_RINGERON;
+ break;
+ case DAHDI_EVENT_RINGEROFF:
+ res = ANALOG_EVENT_RINGEROFF;
+ break;
+ case DAHDI_EVENT_RINGBEGIN:
+ res = ANALOG_EVENT_RINGBEGIN;
+ break;
+ case DAHDI_EVENT_PULSE_START:
+ res = ANALOG_EVENT_PULSE_START;
+ break;
+ case DAHDI_EVENT_NEONMWI_ACTIVE:
+ res = ANALOG_EVENT_NEONMWI_ACTIVE;
+ break;
+ case DAHDI_EVENT_NEONMWI_INACTIVE:
+ res = ANALOG_EVENT_NEONMWI_INACTIVE;
+ break;
}
+/* Note by jpeeler: This function has a rather large section of code ifdefed
+ * away. I'd like to leave the code there until more testing is done and I
+ * know for sure that nothing got left out. The plan is at the latest for this
+ * comment and code below to be removed shortly after the merging of sig_pri.
+ */
static void *__analog_ss_thread(void *data)
{
struct analog_pvt *p = data;
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
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