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-rw-r--r--net/asterisk16/Makefile18
-rw-r--r--net/asterisk16/distinfo12
-rw-r--r--net/asterisk16/files/codecnego-patch-Makefile28
-rw-r--r--net/asterisk16/files/dtmf_debug.diff225
-rw-r--r--net/asterisk16/files/feature_disconnect.diff292
-rw-r--r--net/asterisk16/files/nocodecnego-patch-Makefile28
-rw-r--r--net/asterisk16/files/patch-apps::app_dial.c88
-rw-r--r--net/asterisk16/files/patch-channels::chan_sip.c25
-rw-r--r--net/asterisk16/files/patch-channels::h323::ast_h323.cxx29
-rw-r--r--net/asterisk16/files/patch-include::asterisk::features.h57
-rw-r--r--net/asterisk16/files/patch-main::Makefile7
-rw-r--r--net/asterisk16/files/patch-main::manager.c31
-rw-r--r--net/asterisk16/files/patch-res::res_features.c144
-rw-r--r--net/asterisk16/files/rtp_force_dtmf-codecnego.diff86
-rw-r--r--net/asterisk16/files/rtp_force_dtmf-nocodecnego.diff70
-rw-r--r--net/asterisk16/files/sip_force_callid.diff12
-rw-r--r--net/asterisk16/pkg-plist22
17 files changed, 822 insertions, 352 deletions
diff --git a/net/asterisk16/Makefile b/net/asterisk16/Makefile
index f972ca541c79..c1c8e5eaa5c1 100644
--- a/net/asterisk16/Makefile
+++ b/net/asterisk16/Makefile
@@ -6,8 +6,7 @@
#
PORTNAME= asterisk
-PORTVERSION= 1.4.18.1
-PORTREVISION= 1
+PORTVERSION= 1.4.19.2
CATEGORIES= net
MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
http://ftp.digium.com/pub/asterisk/old-releases/
@@ -50,7 +49,8 @@ OPTIONS= OGGVORBIS "Enable Ogg Vorbis support" on \
FREETDS "Enable FreeTDS support" on \
JABBER "Enable Jabber and Gtalk support" on \
SQLITE "Enable SQLITE support" on \
- CODEC_PATCH "Apply codec negotiation patch" off
+ CODEC_PATCH "Apply codec negotiation patch" off \
+ MISC_PATCHES "Apply additional patches" off
.include <bsd.port.pre.mk>
@@ -159,13 +159,23 @@ LIB_DEPENDS+= sqlite:${PORTSDIR}/databases/sqlite2
.endif
.if defined(WITH_CODEC_PATCH)
-PATCHFILES= asterisk-1.4.18-codec-negotiation-20080313.diff.gz
+PATCHFILES= asterisk-1.4.19-codec-negotiation-20080408.diff.gz
PATCH_SITES= http://b2bua.org/chrome/site/
EXTRA_PATCHES+= ${PATCHDIR}/codecnego-patch-Makefile
.else
EXTRA_PATCHES+= ${PATCHDIR}/nocodecnego-patch-Makefile
.endif
+.if defined(WITH_MISC_PATCHES)
+EXTRA_PATCHES+= ${PATCHDIR}/dtmf_debug.diff
+EXTRA_PATCHES+= ${PATCHDIR}/feature_disconnect.diff
+EXTRA_PATCHES+= ${PATCHDIR}/sip_force_callid.diff
+.if defined(WITH_CODEC_PATCH)
+EXTRA_PATCHES+= ${PATCHDIR}/rtp_force_dtmf-codecnego.diff
+.else
+EXTRA_PATCHES+= ${PATCHDIR}/rtp_force_dtmf-nocodecnego.diff
+.endif
+.endif
post-patch:
${REINPLACE_CMD} -e 's|/var/lib|${PREFIX}/share|g' ${WRKSRC}/configs/musiconhold.conf.sample
diff --git a/net/asterisk16/distinfo b/net/asterisk16/distinfo
index c65a3a6dbea6..64a1135cfcd3 100644
--- a/net/asterisk16/distinfo
+++ b/net/asterisk16/distinfo
@@ -1,6 +1,6 @@
-MD5 (asterisk-1.4.18.1.tar.gz) = a0d7ee4054a04529b745e60dd4e750c4
-SHA256 (asterisk-1.4.18.1.tar.gz) = 5ed2e8b66fc8ac96e516b0ddfea065012e39ef51c8f35ed69c6d7cd74e180d78
-SIZE (asterisk-1.4.18.1.tar.gz) = 11488923
-MD5 (asterisk-1.4.18-codec-negotiation-20080313.diff.gz) = c2bc3b82217383936961a61b239e4635
-SHA256 (asterisk-1.4.18-codec-negotiation-20080313.diff.gz) = 6a6ec9436de24e8c35d5922f7116e65e08052bc6bef6ecc12790736bcf058af5
-SIZE (asterisk-1.4.18-codec-negotiation-20080313.diff.gz) = 38966
+MD5 (asterisk-1.4.19.2.tar.gz) = a942367e20a364510881c086a3517321
+SHA256 (asterisk-1.4.19.2.tar.gz) = b3248b84ec824c3c21861607fdc0c71e4acee16edc570004ee6c9e2f712705c4
+SIZE (asterisk-1.4.19.2.tar.gz) = 11496546
+MD5 (asterisk-1.4.19-codec-negotiation-20080408.diff.gz) = 12db6a4bc354d3eeeee7a900a6060b2f
+SHA256 (asterisk-1.4.19-codec-negotiation-20080408.diff.gz) = 55290639455b1117541c5bf560ef8326c3e9414d266459293893b0e3d36c64f9
+SIZE (asterisk-1.4.19-codec-negotiation-20080408.diff.gz) = 39004
diff --git a/net/asterisk16/files/codecnego-patch-Makefile b/net/asterisk16/files/codecnego-patch-Makefile
index f474578c3a35..c94aad32f623 100644
--- a/net/asterisk16/files/codecnego-patch-Makefile
+++ b/net/asterisk16/files/codecnego-patch-Makefile
@@ -1,6 +1,6 @@
---- Makefile.orig Mon Dec 24 11:53:33 2007
-+++ Makefile Mon Dec 24 11:53:33 2007
-@@ -127,7 +127,7 @@
+--- Makefile.orig 2008-04-08 14:50:57.000000000 +0300
++++ Makefile 2008-04-08 14:52:01.000000000 +0300
+@@ -129,7 +129,7 @@
ASTMANDIR=$(mandir)
ifneq ($(findstring BSD,$(OSARCH)),)
ASTVARLIBDIR=$(prefix)/share/asterisk
@@ -9,7 +9,7 @@
else
ASTVARLIBDIR=$(localstatedir)/lib/asterisk
endif
-@@ -222,7 +222,7 @@
+@@ -224,7 +224,7 @@
ifeq ($(OSARCH),FreeBSD)
# -V is understood by BSD Make, not by GNU make.
@@ -18,7 +18,7 @@
ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
AST_LIBS+=$(shell if test $(BSDVERSION) -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
endif
-@@ -402,15 +402,15 @@
+@@ -404,15 +404,15 @@
# Should static HTTP be installed during make samples or even with its own target ala
# webvoicemail? There are portions here that *could* be customized but might also be
# improved a lot. I'll put it here for now.
@@ -39,7 +39,7 @@
$(MAKE) -C sounds install
update:
-@@ -431,45 +431,45 @@
+@@ -433,45 +433,45 @@
OLDHEADERS=$(filter-out $(NEWHEADERS),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
bininstall: _all
@@ -115,16 +115,16 @@
fi
$(SUBDIRS_INSTALL):
-@@ -496,7 +496,7 @@
- echo " WARNING WARNING WARNING" ;\
- fi
+@@ -505,7 +505,7 @@
+ @exit 1
+ endif
--install: datafiles bininstall $(SUBDIRS_INSTALL)
-+install: datafiles bininstall $(SUBDIRS_INSTALL) samples
+-install: badshell datafiles bininstall $(SUBDIRS_INSTALL)
++install: badshell datafiles bininstall $(SUBDIRS_INSTALL) samples
@if [ -x /usr/sbin/asterisk-post-install ]; then \
/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
fi
-@@ -534,31 +534,23 @@
+@@ -543,31 +543,23 @@
upgrade: bininstall
adsi:
@@ -164,7 +164,7 @@
( \
echo "[directories]" ; \
echo "astetcdir => $(ASTETCDIR)" ; \
-@@ -643,20 +635,23 @@
+@@ -653,20 +645,23 @@
echo "; Default: strict"; \
echo ";"; \
echo ";translation_algorithm = strict"; \
@@ -192,7 +192,7 @@
done
@echo " +--------- Asterisk Web Voicemail ----------+"
@echo " + +"
-@@ -683,10 +678,10 @@
+@@ -693,10 +688,10 @@
__rpm: include/asterisk/version.h include/asterisk/buildopts.h spec
rm -rf /tmp/asterisk ; \
diff --git a/net/asterisk16/files/dtmf_debug.diff b/net/asterisk16/files/dtmf_debug.diff
new file mode 100644
index 000000000000..81205d40dd46
--- /dev/null
+++ b/net/asterisk16/files/dtmf_debug.diff
@@ -0,0 +1,225 @@
+--- channels/chan_sip.c.orig 2008-03-18 16:42:59.000000000 +0200
++++ channels/chan_sip.c 2008-03-18 17:08:34.000000000 +0200
+@@ -3768,6 +3768,7 @@
+ ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
+ else {
+ p->owner = newchan;
++ ast_rtp_set_chan_name(p->rtp, newchan->name);
+ ret = 0;
+ }
+ if (option_debug > 2)
+@@ -4032,6 +4035,7 @@
+ if (i->rtp) {
+ tmp->fds[0] = ast_rtp_fd(i->rtp);
+ tmp->fds[1] = ast_rtcp_fd(i->rtp);
++ ast_rtp_set_chan_id(i->rtp, i->callid);
+ }
+ if (needvideo && i->vrtp) {
+ tmp->fds[2] = ast_rtp_fd(i->vrtp);
+@@ -4059,6 +4063,8 @@
+ if (!ast_strlen_zero(i->language))
+ ast_string_field_set(tmp, language, i->language);
+ i->owner = tmp;
++ ast_rtp_set_chan_name(i->rtp, tmp->name);
++
+ ast_module_ref(ast_module_info->self);
+ ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
+ /*Since it is valid to have extensions in the dialplan that have unescaped characters in them
+@@ -4479,8 +4485,10 @@
+ build_via(p);
+ if (!callid)
+ build_callid_pvt(p);
+- else
++ else {
+ ast_string_field_set(p, callid, callid);
++ ast_rtp_set_chan_id(p->rtp, p->callid);
++ }
+ /* Assign default music on hold class */
+ ast_string_field_set(p, mohinterpret, default_mohinterpret);
+ ast_string_field_set(p, mohsuggest, default_mohsuggest);
+--- include/asterisk/rtp.h.orig 2008-03-18 13:35:42.000000000 +0200
++++ include/asterisk/rtp.h 2008-03-18 13:35:58.000000000 +0200
+@@ -251,6 +251,9 @@
+
+ int ast_rtp_codec_getformat(int pt);
+
++void ast_rtp_set_chan_name(struct ast_rtp *, const char *);
++void ast_rtp_set_chan_id(struct ast_rtp *, const char *);
++
+ /*! \brief Set rtp timeout */
+ void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
+ /*! \brief Set rtp hold timeout */
+--- main/rtp.c.orig 2008-04-08 14:53:18.000000000 +0300
++++ main/rtp.c 2008-04-08 14:54:14.000000000 +0300
+@@ -81,6 +81,7 @@
+ static int rtpstart; /*!< First port for RTP sessions (set in rtp.conf) */
+ static int rtpend; /*!< Last port for RTP sessions (set in rtp.conf) */
+ static int rtpdebug; /*!< Are we debugging? */
++static int rtpdebugdtmf; /*!< Are we debugging DTMFs? */
+ static int rtcpdebug; /*!< Are we debugging RTCP? */
+ static int rtcpstats; /*!< Are we debugging RTCP? */
+ static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
+@@ -168,6 +169,8 @@
+ struct ast_codec_pref pref;
+ struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
+ int set_marker_bit:1; /*!< Whether to set the marker bit or not */
++ char chan_name[100];
++ char chan_id[100];
+ };
+
+ /* Forward declarations */
+@@ -669,8 +672,8 @@
+ struct ast_frame *f = NULL;
+ event = ntohl(*((unsigned int *)(data)));
+ event &= 0x001F;
+- if (option_debug > 2 || rtpdebug)
+- ast_log(LOG_DEBUG, "Cisco DTMF Digit: %08x (len = %d)\n", event, len);
++ if (option_debug > 2 || rtpdebug || rtpdebugdtmf)
++ ast_log(LOG_DEBUG, "Channel: %s %s Cisco DTMF packet: %08x (len = %d)\n", rtp->chan_name, rtp->chan_id, event, len);
+ if (event < 10) {
+ resp = '0' + event;
+ } else if (event < 11) {
+@@ -684,12 +687,24 @@
+ }
+ if (rtp->resp && (rtp->resp != resp)) {
+ f = send_dtmf(rtp, AST_FRAME_DTMF_END);
++ ast_log(LOG_DEBUG, "Channel: %s %s Cisco DTMF event: %c\n", rtp->chan_name, rtp->chan_id, rtp->resp);
+ }
+ rtp->resp = resp;
+ rtp->dtmfcount = dtmftimeout;
+ return f;
+ }
+
++void ast_rtp_set_chan_id(struct ast_rtp *rtp, const char *chan_id) {
++ if (rtp == NULL || chan_id == NULL)
++ return;
++ snprintf(rtp->chan_id, sizeof(rtp->chan_id), "%s", chan_id);
++}
++
++void ast_rtp_set_chan_name(struct ast_rtp *rtp, const char *chan_name) {
++ if (rtp == NULL || chan_name == NULL)
++ return;
++ snprintf(rtp->chan_name, sizeof(rtp->chan_name), "%s", chan_name);
++}
+ /*!
+ * \brief Process RTP DTMF and events according to RFC 2833.
+ *
+@@ -1051,6 +1066,10 @@
+ struct rtpPayloadType rtpPT;
+ int reconstruct = ntohl(rtpheader[0]);
+
++ /* If we are listening for DTMF - then feed all packets into the core to keep the RTP stream consistent when relaying DTMFs */
++ if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF))
++ return -1;
++
+ /* Get fields from packet */
+ payload = (reconstruct & 0x7f0000) >> 16;
+ mark = (((reconstruct & 0x800000) >> 23) != 0);
+@@ -1062,10 +1081,6 @@
+ if (!bridged->current_RTP_PT[payload].code)
+ return -1;
+
+- /* If the payload is DTMF, and we are listening for DTMF - then feed it into the core */
+- if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) && !rtpPT.isAstFormat && rtpPT.code == AST_RTP_DTMF)
+- return -1;
+-
+ /* Otherwise adjust bridged payload to match */
+ bridged_payload = ast_rtp_lookup_code(bridged, rtpPT.isAstFormat, rtpPT.code);
+
+@@ -1254,11 +1269,12 @@
+ /* This is special in-band data that's not one of our codecs */
+ if (rtpPT.code == AST_RTP_DTMF) {
+ /* It's special -- rfc2833 process it */
+- if (rtp_debug_test_addr(&sin)) {
++ if (rtp_debug_test_addr(&sin) || rtpdebugdtmf) {
+ unsigned char *data;
+ unsigned int event;
+ unsigned int event_end;
+ unsigned int duration;
++
+ data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
+ event = ntohl(*((unsigned int *)(data)));
+ event >>= 24;
+@@ -1267,9 +1283,12 @@
+ event_end >>= 24;
+ duration = ntohl(*((unsigned int *)(data)));
+ duration &= 0xFFFF;
+- ast_verbose("Got RTP RFC2833 from %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
++
++ ast_verbose("Channel: %s %s Got RTP RFC2833 from %s:%u to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d)\n", rtp->chan_name, rtp->chan_id, ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), ast_inet_ntoa(rtp->us.sin_addr), ntohs(rtp->us.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
+ }
+ f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp);
++ if (rtpdebugdtmf && f)
++ ast_verbose("Channel: %s %s Got RFC2833 DTMF event %c of type %s\n", rtp->chan_name, rtp->chan_id, f->subclass, (f->frametype == AST_FRAME_DTMF_BEGIN ? "DTMF BEGIN" : (f->frametype == AST_FRAME_DTMF_END ? "DTMF_END" : "UNKNOWN")));
+ } else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
+ /* It's really special -- process it the Cisco way */
+ if (rtp->lastevent <= seqno || (rtp->lastevent >= 65530 && seqno <= 6)) {
+@@ -2198,8 +2217,9 @@
+ ast_log(LOG_ERROR, "RTP Transmission error to %s:%u: %s\n",
+ ast_inet_ntoa(rtp->them.sin_addr),
+ ntohs(rtp->them.sin_port), strerror(errno));
+- if (rtp_debug_test_addr(&rtp->them))
+- ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++ if (rtp_debug_test_addr(&rtp->them) || rtpdebugdtmf)
++ ast_verbose("Channel: %s %s Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++ rtp->chan_name, rtp->chan_id,
+ ast_inet_ntoa(rtp->them.sin_addr),
+ ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+ /* Increment sequence number */
+@@ -2242,8 +2262,9 @@
+ ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
+ ast_inet_ntoa(rtp->them.sin_addr),
+ ntohs(rtp->them.sin_port), strerror(errno));
+- if (rtp_debug_test_addr(&rtp->them))
+- ast_verbose("Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++ if (rtp_debug_test_addr(&rtp->them) || rtpdebugdtmf)
++ ast_verbose("Channel: %s %s Sent RTP DTMF packet to %s:%u (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
++ rtp->chan_name, rtp->chan_id,
+ ast_inet_ntoa(rtp->them.sin_addr),
+ ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+
+@@ -3481,6 +3502,16 @@
+ return RESULT_SUCCESS;
+ }
+
++static int rtp_do_debug_dtmf(int fd, int argc, char *argv[])
++{
++ if (argc != 3)
++ return RESULT_SHOWUSAGE;
++
++ rtpdebugdtmf = 1;
++ ast_cli(fd, "RTP DTMF debugging enabled\n");
++ return RESULT_SUCCESS;
++}
++
+ static int rtp_do_debug(int fd, int argc, char *argv[])
+ {
+ if (argc != 2) {
+@@ -3541,6 +3572,7 @@
+ if (argc != 3)
+ return RESULT_SHOWUSAGE;
+ rtpdebug = 0;
++ rtpdebugdtmf = 0;
+ ast_cli(fd,"RTP Debugging Disabled\n");
+ return RESULT_SUCCESS;
+ }
+@@ -3601,7 +3633,7 @@
+ }
+
+ static char debug_usage[] =
+- "Usage: rtp debug [ip host[:port]]\n"
++ "Usage: rtp debug [ip host[:port] | dtmf]\n"
+ " Enable dumping of all RTP packets to and from host.\n";
+
+ static char no_debug_usage[] =
+@@ -3676,6 +3708,10 @@
+ rtp_do_debug, "Enable RTP debugging",
+ debug_usage },
+
++ { { "rtp", "debug", "dtmf", NULL },
++ rtp_do_debug_dtmf, "Enable RTP debugging on DTMFs",
++ debug_usage },
++
+ { { "rtp", "debug", "off", NULL },
+ rtp_no_debug, "Disable RTP debugging",
+ no_debug_usage, NULL, &cli_rtp_no_debug_deprecated },
diff --git a/net/asterisk16/files/feature_disconnect.diff b/net/asterisk16/files/feature_disconnect.diff
new file mode 100644
index 000000000000..5be713b0da13
--- /dev/null
+++ b/net/asterisk16/files/feature_disconnect.diff
@@ -0,0 +1,292 @@
+--- apps/app_dial.c.orig Wed Feb 6 18:45:30 2008
++++ apps/app_dial.c Wed Feb 6 18:51:59 2008
+@@ -297,6 +297,8 @@ AST_APP_OPTIONS(dial_exec_options, {
+ OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
+ OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK))
+
++static int detect_disconnect(struct ast_channel *chan, char code);
++
+ /* We define a custom "local user" structure because we
+ use it not only for keeping track of what is in use but
+ also for keeping track of who we're dialing. */
+@@ -402,6 +404,7 @@ static struct ast_channel *wait_for_answ
+ struct ast_channel *peer = NULL;
+ /* single is set if only one destination is enabled */
+ int single = outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
++
+
+ if (single) {
+ /* Turn off hold music, etc */
+@@ -716,9 +719,9 @@ static struct ast_channel *wait_for_answ
+ }
+
+ if (ast_test_flag(peerflags, OPT_CALLER_HANGUP) &&
+- (f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
++ detect_disconnect(in, f->subclass)) { /* hmm it it not guaranteed to be '*' anymore. */
+ if (option_verbose > 2)
+- ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
++ ast_verbose(VERBOSE_PREFIX_3 "User requested call disconnect.\n");
+ *to=0;
+ ast_cdr_noanswer(in->cdr);
+ strcpy(status, "CANCEL");
+@@ -776,6 +779,56 @@ static struct ast_channel *wait_for_answ
+
+ return peer;
+ }
++
++static char featurecode[FEATURE_MAX_LEN + 1] = "";
++
++static int detect_disconnect(struct ast_channel *chan, char code)
++{
++ struct feature_interpret_result result;
++ int x;
++ struct ast_flags features;
++ int res = FEATURE_RETURN_PASSDIGITS;
++ struct ast_call_feature *feature;
++ char *cptr;
++ int len;
++
++ len = strlen(featurecode);
++ if (len >= FEATURE_MAX_LEN) {
++ featurecode[0] = '\0';
++ }
++ cptr = &featurecode[strlen(featurecode)];
++ cptr[0] = code;
++ cptr[1] = '\0';
++
++ memset(&features, 0, sizeof(struct ast_flags));
++ ast_set_flag(&features, AST_FEATURE_DISCONNECT);
++
++ ast_features_lock();
++
++ res = ast_feature_detect(chan, &features, featurecode, &result);
++
++ if (res != FEATURE_RETURN_STOREDIGITS)
++ featurecode[0] = '\0';
++
++ if (result.builtin_feature && result.builtin_feature->feature_mask & AST_FEATURE_DISCONNECT) {
++ ast_features_unlock();
++ return 1;
++ }
++
++ for (x = 0; x < result.num_dyn_features; ++x) {
++ feature = result.dynamic_features[x];
++ if (feature->feature_mask & AST_FEATURE_DISCONNECT) {
++ ast_features_unlock();
++ return 1;
++ }
++ }
++
++ ast_features_unlock();
++
++ return 0;
++}
++
++
+
+ static void replace_macro_delimiter(char *s)
+ {
+--- include/asterisk/features.h.orig Thu Aug 23 23:16:41 2007
++++ include/asterisk/features.h Tue Dec 11 17:13:52 2007
+@@ -31,6 +31,20 @@
+ #define FEATURE_EXTEN_LEN 32
+ #define FEATURE_MOH_LEN 80 /* same as MAX_MUSICCLASS from channel.h */
+
++#define FEATURE_RETURN_HANGUP -1
++#define FEATURE_RETURN_SUCCESSBREAK 0
++#define FEATURE_RETURN_PBX_KEEPALIVE AST_PBX_KEEPALIVE
++#define FEATURE_RETURN_NO_HANGUP_PEER AST_PBX_NO_HANGUP_PEER
++#define FEATURE_RETURN_PASSDIGITS 21
++#define FEATURE_RETURN_STOREDIGITS 22
++#define FEATURE_RETURN_SUCCESS 23
++#define FEATURE_RETURN_KEEPTRYING 24
++
++#define FEATURE_SENSE_CHAN (1 << 0)
++#define FEATURE_SENSE_PEER (1 << 1)
++
++typedef int (*feature_operation)(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data);
++
+ /*! \brief main call feature structure */
+ struct ast_call_feature {
+ int feature_mask;
+@@ -38,7 +52,7 @@
+ char sname[FEATURE_SNAME_LEN];
+ char exten[FEATURE_MAX_LEN];
+ char default_exten[FEATURE_MAX_LEN];
+- int (*operation)(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data);
++ feature_operation operation;
+ unsigned int flags;
+ char app[FEATURE_APP_LEN];
+ char app_args[FEATURE_APP_ARGS_LEN];
+@@ -47,6 +61,12 @@
+ };
+
+
++struct feature_interpret_result {
++ struct ast_call_feature *builtin_feature;
++ struct ast_call_feature *dynamic_features[20];
++ int num_dyn_features;
++};
++
+
+ /*! \brief Park a call and read back parked location
+ * \param chan the channel to actually be parked
+@@ -93,5 +113,11 @@
+ /*! \brief unregister feature from feature_set
+ \param feature the ast_call_feature object which was registered before*/
+ void ast_unregister_feature(struct ast_call_feature *feature);
++
++int ast_feature_detect(struct ast_channel *chan, const struct ast_flags *features, char *code, struct feature_interpret_result *result);
++
++void ast_features_lock(void);
++void ast_features_unlock(void);
++
+
+ #endif /* _AST_FEATURES_H */
+--- res/res_features.c.orig 2008-04-08 14:55:25.000000000 +0300
++++ res/res_features.c 2008-04-08 14:59:59.000000000 +0300
+@@ -485,18 +485,6 @@
+ }
+
+
+-#define FEATURE_RETURN_HANGUP -1
+-#define FEATURE_RETURN_SUCCESSBREAK 0
+-#define FEATURE_RETURN_PBX_KEEPALIVE AST_PBX_KEEPALIVE
+-#define FEATURE_RETURN_NO_HANGUP_PEER AST_PBX_NO_HANGUP_PEER
+-#define FEATURE_RETURN_PASSDIGITS 21
+-#define FEATURE_RETURN_STOREDIGITS 22
+-#define FEATURE_RETURN_SUCCESS 23
+-#define FEATURE_RETURN_KEEPTRYING 24
+-
+-#define FEATURE_SENSE_CHAN (1 << 0)
+-#define FEATURE_SENSE_PEER (1 << 1)
+-
+ /*! \brief
+ * set caller and callee according to the direction
+ */
+@@ -1061,32 +1049,35 @@
+ return res;
+ }
+
+-static int ast_feature_interpret(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense)
++void ast_features_lock(void)
++{
++ ast_rwlock_rdlock(&features_lock);
++ AST_LIST_LOCK(&feature_list);
++}
++
++void ast_features_unlock(void)
++{
++ AST_LIST_UNLOCK(&feature_list);
++ ast_rwlock_unlock(&features_lock);
++}
++
++int ast_feature_detect(struct ast_channel *chan, const struct ast_flags *features, char *code, struct feature_interpret_result *result)
+ {
+ int x;
+- struct ast_flags features;
+ int res = FEATURE_RETURN_PASSDIGITS;
+ struct ast_call_feature *feature;
+- const char *dynamic_features;
++ const char *dynamic_features = pbx_builtin_getvar_helper(chan, "DYNAMIC_FEATURES");
+ char *tmp, *tok;
+
+- if (sense == FEATURE_SENSE_CHAN) {
+- ast_copy_flags(&features, &(config->features_caller), AST_FLAGS_ALL);
+- dynamic_features = pbx_builtin_getvar_helper(chan, "DYNAMIC_FEATURES");
+- } else {
+- ast_copy_flags(&features, &(config->features_callee), AST_FLAGS_ALL);
+- dynamic_features = pbx_builtin_getvar_helper(peer, "DYNAMIC_FEATURES");
+- }
+- if (option_debug > 2)
+- ast_log(LOG_DEBUG, "Feature interpret: chan=%s, peer=%s, code=%s, sense=%d, features=%d dynamic=%s\n", chan->name, peer->name, code, sense, features.flags, dynamic_features);
++ result->builtin_feature = NULL;
++ result->num_dyn_features = 0;
+
+- ast_rwlock_rdlock(&features_lock);
+ for (x = 0; x < FEATURES_COUNT; x++) {
+- if ((ast_test_flag(&features, builtin_features[x].feature_mask)) &&
++ if ((ast_test_flag(features, builtin_features[x].feature_mask)) &&
+ !ast_strlen_zero(builtin_features[x].exten)) {
+ /* Feature is up for consideration */
+ if (!strcmp(builtin_features[x].exten, code)) {
+- res = builtin_features[x].operation(chan, peer, config, code, sense, NULL);
++ result->builtin_feature = &builtin_features[x];
+ break;
+ } else if (!strncmp(builtin_features[x].exten, code, strlen(code))) {
+ if (res == FEATURE_RETURN_PASSDIGITS)
+@@ -1094,7 +1085,6 @@
+ }
+ }
+ }
+- ast_rwlock_unlock(&features_lock);
+
+ if (ast_strlen_zero(dynamic_features))
+ return res;
+@@ -1102,9 +1092,7 @@
+ tmp = ast_strdupa(dynamic_features);
+
+ while ((tok = strsep(&tmp, "#"))) {
+- AST_LIST_LOCK(&feature_list);
+ if (!(feature = find_dynamic_feature(tok))) {
+- AST_LIST_UNLOCK(&feature_list);
+ continue;
+ }
+
+@@ -1112,21 +1100,52 @@
+ if (!strcmp(feature->exten, code)) {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 " Feature Found: %s exten: %s\n",feature->sname, tok);
+- res = feature->operation(chan, peer, config, code, sense, feature);
+- if (res != FEATURE_RETURN_KEEPTRYING) {
+- AST_LIST_UNLOCK(&feature_list);
++ result->dynamic_features[result->num_dyn_features++] = feature;
++ if (result->num_dyn_features >= (sizeof(result->dynamic_features) / sizeof(result->dynamic_features[0]))) {
+ break;
+ }
+ res = FEATURE_RETURN_PASSDIGITS;
+ } else if (!strncmp(feature->exten, code, strlen(code)))
+ res = FEATURE_RETURN_STOREDIGITS;
+
+- AST_LIST_UNLOCK(&feature_list);
+ }
+
+ return res;
+ }
+
++static int ast_feature_interpret(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense)
++{
++ struct feature_interpret_result result;
++ int x;
++ struct ast_flags features;
++ int res = FEATURE_RETURN_PASSDIGITS;
++ struct ast_call_feature *feature;
++
++ if (sense == FEATURE_SENSE_CHAN)
++ ast_copy_flags(&features, &(config->features_caller), AST_FLAGS_ALL);
++ else
++ ast_copy_flags(&features, &(config->features_callee), AST_FLAGS_ALL);
++ if (option_debug > 2)
++ ast_log(LOG_DEBUG, "Feature interpret: chan=%s, peer=%s, sense=%d, features=%d\n", chan->name, peer->name, sense, features.flags);
++
++ ast_features_lock();
++ res = ast_feature_detect(chan, &features, code, &result);
++
++ if (result.builtin_feature)
++ res = result.builtin_feature->operation(chan, peer, config, code, sense, NULL);
++
++ for (x = 0; x < result.num_dyn_features; ++x) {
++ feature = result.dynamic_features[x];
++ res = feature->operation(chan, peer, config, code, sense, feature);
++ if (res != FEATURE_RETURN_KEEPTRYING)
++ break;
++ res = FEATURE_RETURN_PASSDIGITS;
++ }
++
++ ast_features_unlock();
++ return res;
++}
++
+ static void set_config_flags(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config)
+ {
+ int x;
diff --git a/net/asterisk16/files/nocodecnego-patch-Makefile b/net/asterisk16/files/nocodecnego-patch-Makefile
index f73c3d83bcde..1d9675f1d664 100644
--- a/net/asterisk16/files/nocodecnego-patch-Makefile
+++ b/net/asterisk16/files/nocodecnego-patch-Makefile
@@ -1,6 +1,6 @@
---- Makefile.orig Mon Dec 17 08:44:51 2007
-+++ Makefile Mon Dec 24 11:54:52 2007
-@@ -127,7 +127,7 @@
+--- Makefile.orig 2008-03-25 16:37:35.000000000 +0200
++++ Makefile 2008-04-08 15:02:12.000000000 +0300
+@@ -129,7 +129,7 @@
ASTMANDIR=$(mandir)
ifneq ($(findstring BSD,$(OSARCH)),)
ASTVARLIBDIR=$(prefix)/share/asterisk
@@ -9,7 +9,7 @@
else
ASTVARLIBDIR=$(localstatedir)/lib/asterisk
endif
-@@ -222,7 +222,7 @@
+@@ -224,7 +224,7 @@
ifeq ($(OSARCH),FreeBSD)
# -V is understood by BSD Make, not by GNU make.
@@ -18,7 +18,7 @@
ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
AST_LIBS+=$(shell if test $(BSDVERSION) -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
endif
-@@ -402,15 +402,15 @@
+@@ -404,15 +404,15 @@
# Should static HTTP be installed during make samples or even with its own target ala
# webvoicemail? There are portions here that *could* be customized but might also be
# improved a lot. I'll put it here for now.
@@ -39,7 +39,7 @@
$(MAKE) -C sounds install
update:
-@@ -431,45 +431,45 @@
+@@ -433,45 +433,45 @@
OLDHEADERS=$(filter-out $(NEWHEADERS),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
bininstall: _all
@@ -115,16 +115,16 @@
fi
$(SUBDIRS_INSTALL):
-@@ -496,7 +496,7 @@
- echo " WARNING WARNING WARNING" ;\
- fi
+@@ -505,7 +505,7 @@
+ @exit 1
+ endif
--install: datafiles bininstall $(SUBDIRS_INSTALL)
-+install: datafiles bininstall $(SUBDIRS_INSTALL) samples
+-install: badshell datafiles bininstall $(SUBDIRS_INSTALL)
++install: badshell datafiles bininstall $(SUBDIRS_INSTALL) samples
@if [ -x /usr/sbin/asterisk-post-install ]; then \
/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
fi
-@@ -534,31 +534,23 @@
+@@ -543,31 +543,23 @@
upgrade: bininstall
adsi:
@@ -164,7 +164,7 @@
( \
echo "[directories]" ; \
echo "astetcdir => $(ASTETCDIR)" ; \
-@@ -602,20 +594,23 @@
+@@ -612,20 +604,23 @@
echo ";astctlowner = root" ; \
echo ";astctlgroup = apache" ; \
echo ";astctl = asterisk.ctl" ; \
@@ -192,7 +192,7 @@
done
@echo " +--------- Asterisk Web Voicemail ----------+"
@echo " + +"
-@@ -642,10 +637,10 @@
+@@ -652,10 +647,10 @@
__rpm: include/asterisk/version.h include/asterisk/buildopts.h spec
rm -rf /tmp/asterisk ; \
diff --git a/net/asterisk16/files/patch-apps::app_dial.c b/net/asterisk16/files/patch-apps::app_dial.c
deleted file mode 100644
index 6abe48a17d9f..000000000000
--- a/net/asterisk16/files/patch-apps::app_dial.c
+++ /dev/null
@@ -1,88 +0,0 @@
---- apps/app_dial.c.orig Wed Feb 6 18:45:30 2008
-+++ apps/app_dial.c Wed Feb 6 18:51:59 2008
-@@ -297,6 +297,8 @@ AST_APP_OPTIONS(dial_exec_options, {
- OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
- OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK))
-
-+static int detect_disconnect(struct ast_channel *chan, char code);
-+
- /* We define a custom "local user" structure because we
- use it not only for keeping track of what is in use but
- also for keeping track of who we're dialing. */
-@@ -402,6 +404,7 @@ static struct ast_channel *wait_for_answ
- struct ast_channel *peer = NULL;
- /* single is set if only one destination is enabled */
- int single = outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
-+
-
- if (single) {
- /* Turn off hold music, etc */
-@@ -716,9 +719,9 @@ static struct ast_channel *wait_for_answ
- }
-
- if (ast_test_flag(peerflags, OPT_CALLER_HANGUP) &&
-- (f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
-+ detect_disconnect(in, f->subclass)) { /* hmm it it not guaranteed to be '*' anymore. */
- if (option_verbose > 2)
-- ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
-+ ast_verbose(VERBOSE_PREFIX_3 "User requested call disconnect.\n");
- *to=0;
- ast_cdr_noanswer(in->cdr);
- strcpy(status, "CANCEL");
-@@ -776,6 +779,56 @@ static struct ast_channel *wait_for_answ
-
- return peer;
- }
-+
-+static char featurecode[FEATURE_MAX_LEN + 1] = "";
-+
-+static int detect_disconnect(struct ast_channel *chan, char code)
-+{
-+ struct feature_interpret_result result;
-+ int x;
-+ struct ast_flags features;
-+ int res = FEATURE_RETURN_PASSDIGITS;
-+ struct ast_call_feature *feature;
-+ char *cptr;
-+ int len;
-+
-+ len = strlen(featurecode);
-+ if (len >= FEATURE_MAX_LEN) {
-+ featurecode[0] = '\0';
-+ }
-+ cptr = &featurecode[strlen(featurecode)];
-+ cptr[0] = code;
-+ cptr[1] = '\0';
-+
-+ memset(&features, 0, sizeof(struct ast_flags));
-+ ast_set_flag(&features, AST_FEATURE_DISCONNECT);
-+
-+ ast_features_lock();
-+
-+ res = ast_feature_detect(chan, &features, featurecode, &result);
-+
-+ if (res != FEATURE_RETURN_STOREDIGITS)
-+ featurecode[0] = '\0';
-+
-+ if (result.builtin_feature && result.builtin_feature->feature_mask & AST_FEATURE_DISCONNECT) {
-+ ast_features_unlock();
-+ return 1;
-+ }
-+
-+ for (x = 0; x < result.num_dyn_features; ++x) {
-+ feature = result.dynamic_features[x];
-+ if (feature->feature_mask & AST_FEATURE_DISCONNECT) {
-+ ast_features_unlock();
-+ return 1;
-+ }
-+ }
-+
-+ ast_features_unlock();
-+
-+ return 0;
-+}
-+
-+
-
- static void replace_macro_delimiter(char *s)
- {
diff --git a/net/asterisk16/files/patch-channels::chan_sip.c b/net/asterisk16/files/patch-channels::chan_sip.c
index 916cc7ffa0ac..e54df0eb1bf1 100644
--- a/net/asterisk16/files/patch-channels::chan_sip.c
+++ b/net/asterisk16/files/patch-channels::chan_sip.c
@@ -1,6 +1,6 @@
---- channels/chan_sip.c.orig Mon Dec 24 11:59:46 2007
-+++ channels/chan_sip.c Mon Dec 24 11:58:47 2007
-@@ -493,7 +493,7 @@
+--- channels/chan_sip.c.orig 2008-03-18 16:42:59.000000000 +0200
++++ channels/chan_sip.c 2008-03-18 17:08:34.000000000 +0200
+@@ -495,7 +495,7 @@
#define DEFAULT_MOHINTERPRET "default"
#define DEFAULT_MOHSUGGEST ""
#define DEFAULT_VMEXTEN "asterisk"
@@ -9,16 +9,7 @@
#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
#define DEFAULT_MWITIME 10
#define DEFAULT_ALLOWGUEST TRUE
-@@ -3985,6 +3985,8 @@
- ast_codec_pref_remove2(&tmp->nativeformats, ~i->usercapability);
- fmt = ast_codec_pref_index_audio(&tmp->nativeformats, 0);
-
-+ pbx_builtin_setvar_helper(tmp, "SIP_CODEC_USED", ast_getformatname(fmt));
-+
- /* If we have a prefcodec setting, we have an inbound channel that set a
- preferred format for this call. Otherwise, we check the jointcapability
- We also check for vrtp. If it's not there, we are not allowed do any video anyway.
-@@ -15845,6 +15847,9 @@
+@@ -15873,6 +15881,9 @@
char *ext, *host;
char tmp[256];
char *dest = data;
@@ -26,9 +17,9 @@
+ char *md5secret = NULL;
+ char *authname = NULL;
- if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
- ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", (char *)data);
-@@ -15866,6 +15871,17 @@
+ oldformat = format;
+ if (!(format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1))) {
+@@ -15903,6 +15914,17 @@
if (host) {
*host++ = '\0';
ext = tmp;
@@ -46,7 +37,7 @@
} else {
ext = strchr(tmp, '/');
if (ext)
-@@ -15898,6 +15914,14 @@
+@@ -15933,6 +15955,14 @@
ast_string_field_set(p, username, ext);
ast_string_field_free(p, fullcontact);
}
diff --git a/net/asterisk16/files/patch-channels::h323::ast_h323.cxx b/net/asterisk16/files/patch-channels::h323::ast_h323.cxx
new file mode 100644
index 000000000000..bfc819452128
--- /dev/null
+++ b/net/asterisk16/files/patch-channels::h323::ast_h323.cxx
@@ -0,0 +1,29 @@
+--- channels/h323/ast_h323.cxx.orig 2008-04-08 15:08:37.000000000 +0300
++++ channels/h323/ast_h323.cxx 2008-04-08 15:08:51.000000000 +0300
+@@ -81,8 +81,6 @@
+ /** PWLib entry point */
+ static MyProcess *localProcess = NULL;
+
+-static int _timerChangePipe[2];
+-
+ static unsigned traceOptions = PTrace::Timestamp | PTrace::Thread | PTrace::FileAndLine;
+
+ class PAsteriskLog : public PObject, public iostream {
+@@ -191,8 +189,6 @@
+
+ MyProcess::~MyProcess()
+ {
+- _timerChangePipe[0] = timerChangePipe[0];
+- _timerChangePipe[1] = timerChangePipe[1];
+ }
+
+ void MyProcess::Main()
+@@ -2460,8 +2456,6 @@
+ if (localProcess) {
+ delete localProcess;
+ localProcess = NULL;
+- close(_timerChangePipe[0]);
+- close(_timerChangePipe[1]);
+ }
+ if (logstream) {
+ PTrace::SetLevel(0);
diff --git a/net/asterisk16/files/patch-include::asterisk::features.h b/net/asterisk16/files/patch-include::asterisk::features.h
deleted file mode 100644
index fa5c747eb316..000000000000
--- a/net/asterisk16/files/patch-include::asterisk::features.h
+++ /dev/null
@@ -1,57 +0,0 @@
---- include/asterisk/features.h.orig Thu Aug 23 23:16:41 2007
-+++ include/asterisk/features.h Tue Dec 11 17:13:52 2007
-@@ -31,6 +31,20 @@
- #define FEATURE_EXTEN_LEN 32
- #define FEATURE_MOH_LEN 80 /* same as MAX_MUSICCLASS from channel.h */
-
-+#define FEATURE_RETURN_HANGUP -1
-+#define FEATURE_RETURN_SUCCESSBREAK 0
-+#define FEATURE_RETURN_PBX_KEEPALIVE AST_PBX_KEEPALIVE
-+#define FEATURE_RETURN_NO_HANGUP_PEER AST_PBX_NO_HANGUP_PEER
-+#define FEATURE_RETURN_PASSDIGITS 21
-+#define FEATURE_RETURN_STOREDIGITS 22
-+#define FEATURE_RETURN_SUCCESS 23
-+#define FEATURE_RETURN_KEEPTRYING 24
-+
-+#define FEATURE_SENSE_CHAN (1 << 0)
-+#define FEATURE_SENSE_PEER (1 << 1)
-+
-+typedef int (*feature_operation)(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data);
-+
- /*! \brief main call feature structure */
- struct ast_call_feature {
- int feature_mask;
-@@ -38,7 +52,7 @@
- char sname[FEATURE_SNAME_LEN];
- char exten[FEATURE_MAX_LEN];
- char default_exten[FEATURE_MAX_LEN];
-- int (*operation)(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data);
-+ feature_operation operation;
- unsigned int flags;
- char app[FEATURE_APP_LEN];
- char app_args[FEATURE_APP_ARGS_LEN];
-@@ -47,6 +61,12 @@
- };
-
-
-+struct feature_interpret_result {
-+ struct ast_call_feature *builtin_feature;
-+ struct ast_call_feature *dynamic_features[20];
-+ int num_dyn_features;
-+};
-+
-
- /*! \brief Park a call and read back parked location
- * \param chan the channel to actually be parked
-@@ -93,5 +113,11 @@
- /*! \brief unregister feature from feature_set
- \param feature the ast_call_feature object which was registered before*/
- void ast_unregister_feature(struct ast_call_feature *feature);
-+
-+int ast_feature_detect(struct ast_channel *chan, const struct ast_flags *features, char *code, struct feature_interpret_result *result);
-+
-+void ast_features_lock(void);
-+void ast_features_unlock(void);
-+
-
- #endif /* _AST_FEATURES_H */
diff --git a/net/asterisk16/files/patch-main::Makefile b/net/asterisk16/files/patch-main::Makefile
index 3d1365ea46fa..56624065f61f 100644
--- a/net/asterisk16/files/patch-main::Makefile
+++ b/net/asterisk16/files/patch-main::Makefile
@@ -1,8 +1,5 @@
-
-$FreeBSD$
-
---- main/Makefile.orig
-+++ main/Makefile
+--- main/Makefile.orig 2008-01-29 19:43:41.000000000 +0200
++++ main/Makefile 2008-03-13 13:57:48.000000000 +0200
@@ -102,8 +102,8 @@
cd editline && test -f config.h || CFLAGS="$(PTHREAD_CFLAGS) $(subst $(ASTTOPDIR),../../,$(ASTCFLAGS:-Werror=))" LDFLAGS="$(ASTLDFLAGS)" ./configure --build=$(BUILD_PLATFORM) --host=$(HOST_PLATFORM) --with-ncurses=$(NCURSES_DIR) --with-curses=$(CURSES_DIR) --with-termcap=$(TERMCAP_DIR) --with-tinfo=$(TINFO_DIR)
$(MAKE) -C editline libedit.a
diff --git a/net/asterisk16/files/patch-main::manager.c b/net/asterisk16/files/patch-main::manager.c
new file mode 100644
index 000000000000..686068b09e09
--- /dev/null
+++ b/net/asterisk16/files/patch-main::manager.c
@@ -0,0 +1,31 @@
+--- main/manager.c.orig 2008-04-02 15:49:38.000000000 +0300
++++ main/manager.c 2008-04-02 15:42:18.000000000 +0300
+@@ -176,6 +176,7 @@
+ struct eventqent *eventq;
+ /* Timeout for ast_carefulwrite() */
+ int writetimeout;
++ int pending_event;
+ AST_LIST_ENTRY(mansession) list;
+ };
+
+@@ -2214,6 +2215,11 @@
+ fds[0].events = POLLIN;
+ do {
+ ast_mutex_lock(&s->__lock);
++ if (s->pending_event) {
++ s->pending_event = 0;
++ ast_mutex_unlock(&s->__lock);
++ return 0;
++ }
+ s->waiting_thread = pthread_self();
+ ast_mutex_unlock(&s->__lock);
+
+@@ -2475,6 +2481,8 @@
+ ast_mutex_lock(&s->__lock);
+ if (s->waiting_thread != AST_PTHREADT_NULL)
+ pthread_kill(s->waiting_thread, SIGURG);
++ else
++ s->pending_event = 1;
+ ast_mutex_unlock(&s->__lock);
+ }
+ AST_LIST_UNLOCK(&sessions);
diff --git a/net/asterisk16/files/patch-res::res_features.c b/net/asterisk16/files/patch-res::res_features.c
deleted file mode 100644
index c21775f1690d..000000000000
--- a/net/asterisk16/files/patch-res::res_features.c
+++ /dev/null
@@ -1,144 +0,0 @@
---- res/res_features.c.orig Wed Feb 6 16:55:29 2008
-+++ res/res_features.c Wed Feb 6 17:02:00 2008
-@@ -477,18 +477,6 @@ int ast_masq_park_call(struct ast_channe
- }
-
-
--#define FEATURE_RETURN_HANGUP -1
--#define FEATURE_RETURN_SUCCESSBREAK 0
--#define FEATURE_RETURN_PBX_KEEPALIVE AST_PBX_KEEPALIVE
--#define FEATURE_RETURN_NO_HANGUP_PEER AST_PBX_NO_HANGUP_PEER
--#define FEATURE_RETURN_PASSDIGITS 21
--#define FEATURE_RETURN_STOREDIGITS 22
--#define FEATURE_RETURN_SUCCESS 23
--#define FEATURE_RETURN_KEEPTRYING 24
--
--#define FEATURE_SENSE_CHAN (1 << 0)
--#define FEATURE_SENSE_PEER (1 << 1)
--
- /*! \brief
- * set caller and callee according to the direction
- */
-@@ -1052,32 +1040,35 @@ static int remap_feature(const char *nam
- return res;
- }
-
--static int ast_feature_interpret(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense)
-+void ast_features_lock(void)
-+{
-+ ast_rwlock_rdlock(&features_lock);
-+ AST_LIST_LOCK(&feature_list);
-+}
-+
-+void ast_features_unlock(void)
-+{
-+ AST_LIST_UNLOCK(&feature_list);
-+ ast_rwlock_unlock(&features_lock);
-+}
-+
-+int ast_feature_detect(struct ast_channel *chan, const struct ast_flags *features, char *code, struct feature_interpret_result *result)
- {
- int x;
-- struct ast_flags features;
- int res = FEATURE_RETURN_PASSDIGITS;
- struct ast_call_feature *feature;
-- const char *dynamic_features;
-+ const char *dynamic_features = pbx_builtin_getvar_helper(chan, "DYNAMIC_FEATURES");
- char *tmp, *tok;
-
-- if (sense == FEATURE_SENSE_CHAN) {
-- ast_copy_flags(&features, &(config->features_caller), AST_FLAGS_ALL);
-- dynamic_features = pbx_builtin_getvar_helper(chan, "DYNAMIC_FEATURES");
-- } else {
-- ast_copy_flags(&features, &(config->features_callee), AST_FLAGS_ALL);
-- dynamic_features = pbx_builtin_getvar_helper(peer, "DYNAMIC_FEATURES");
-- }
-- if (option_debug > 2)
-- ast_log(LOG_DEBUG, "Feature interpret: chan=%s, peer=%s, sense=%d, features=%d dynamic=%s\n", chan->name, peer->name, sense, features.flags, dynamic_features);
-+ result->builtin_feature = NULL;
-+ result->num_dyn_features = 0;
-
-- ast_rwlock_rdlock(&features_lock);
- for (x = 0; x < FEATURES_COUNT; x++) {
-- if ((ast_test_flag(&features, builtin_features[x].feature_mask)) &&
-+ if ((ast_test_flag(features, builtin_features[x].feature_mask)) &&
- !ast_strlen_zero(builtin_features[x].exten)) {
- /* Feature is up for consideration */
- if (!strcmp(builtin_features[x].exten, code)) {
-- res = builtin_features[x].operation(chan, peer, config, code, sense, NULL);
-+ result->builtin_feature = &builtin_features[x];
- break;
- } else if (!strncmp(builtin_features[x].exten, code, strlen(code))) {
- if (res == FEATURE_RETURN_PASSDIGITS)
-@@ -1085,7 +1076,6 @@ static int ast_feature_interpret(struct
- }
- }
- }
-- ast_rwlock_unlock(&features_lock);
-
- if (ast_strlen_zero(dynamic_features))
- return res;
-@@ -1093,9 +1083,7 @@ static int ast_feature_interpret(struct
- tmp = ast_strdupa(dynamic_features);
-
- while ((tok = strsep(&tmp, "#"))) {
-- AST_LIST_LOCK(&feature_list);
- if (!(feature = find_dynamic_feature(tok))) {
-- AST_LIST_UNLOCK(&feature_list);
- continue;
- }
-
-@@ -1103,18 +1091,49 @@ static int ast_feature_interpret(struct
- if (!strcmp(feature->exten, code)) {
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 " Feature Found: %s exten: %s\n",feature->sname, tok);
-- res = feature->operation(chan, peer, config, code, sense, feature);
-- if (res != FEATURE_RETURN_KEEPTRYING) {
-- AST_LIST_UNLOCK(&feature_list);
-+ result->dynamic_features[result->num_dyn_features++] = feature;
-+ if (result->num_dyn_features >= (sizeof(result->dynamic_features) / sizeof(result->dynamic_features[0]))) {
- break;
- }
- res = FEATURE_RETURN_PASSDIGITS;
- } else if (!strncmp(feature->exten, code, strlen(code)))
- res = FEATURE_RETURN_STOREDIGITS;
-
-- AST_LIST_UNLOCK(&feature_list);
- }
-
-+ return res;
-+}
-+
-+static int ast_feature_interpret(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense)
-+{
-+ struct feature_interpret_result result;
-+ int x;
-+ struct ast_flags features;
-+ int res = FEATURE_RETURN_PASSDIGITS;
-+ struct ast_call_feature *feature;
-+
-+ if (sense == FEATURE_SENSE_CHAN)
-+ ast_copy_flags(&features, &(config->features_caller), AST_FLAGS_ALL);
-+ else
-+ ast_copy_flags(&features, &(config->features_callee), AST_FLAGS_ALL);
-+ if (option_debug > 2)
-+ ast_log(LOG_DEBUG, "Feature interpret: chan=%s, peer=%s, sense=%d, features=%d\n", chan->name, peer->name, sense, features.flags);
-+
-+ ast_features_lock();
-+ res = ast_feature_detect(chan, &features, code, &result);
-+
-+ if (result.builtin_feature)
-+ res = result.builtin_feature->operation(chan, peer, config, code, sense, NULL);
-+
-+ for (x = 0; x < result.num_dyn_features; ++x) {
-+ feature = result.dynamic_features[x];
-+ res = feature->operation(chan, peer, config, code, sense, feature);
-+ if (res != FEATURE_RETURN_KEEPTRYING)
-+ break;
-+ res = FEATURE_RETURN_PASSDIGITS;
-+ }
-+
-+ ast_features_unlock();
- return res;
- }
-
diff --git a/net/asterisk16/files/rtp_force_dtmf-codecnego.diff b/net/asterisk16/files/rtp_force_dtmf-codecnego.diff
new file mode 100644
index 000000000000..7cd43e08d238
--- /dev/null
+++ b/net/asterisk16/files/rtp_force_dtmf-codecnego.diff
@@ -0,0 +1,86 @@
+--- channels/chan_sip.c.orig 2008-03-12 17:37:00.000000000 +0200
++++ channels/chan_sip.c 2008-03-12 18:17:33.000000000 +0200
+@@ -554,6 +554,9 @@
+ static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
+ static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
+ static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
++static int global_force_dtmf_relay = 0;
++static int global_force_dtmf_relay_pt = 101;
++
+ static int compactheaders; /*!< send compact sip headers */
+ static int recordhistory; /*!< Record SIP history. Off by default */
+ static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
+@@ -4983,6 +4986,8 @@
+ int codec_index = 0;
+ int codec_pt_order[256];
+
++ int dtmf_present = 0;
++
+ if (!p->rtp) {
+ ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
+ return -1;
+@@ -5408,12 +5413,20 @@
+ for (x = 0; x < codec_index; ++x) {
+ struct rtpPayloadType pt;
+ pt = ast_rtp_lookup_pt(p->rtp, codec_pt_order[x]);
++ if (pt.code == AST_RTP_DTMF)
++ dtmf_present = 1;
+ if (!pt.isAstFormat && !pt.code && p->vrtp)
+ pt = ast_rtp_lookup_pt(p->vrtp, codec_pt_order[x]);
+ if (pt.isAstFormat)
+ ast_codec_pref_append(&p->formats, pt.code);
+ }
+ ast_codec_pref_remove2(&p->formats, ~p->usercapability);
++ if (!dtmf_present && global_force_dtmf_relay) {
++ newnoncodeccapability |= AST_RTP_DTMF;
++ ast_rtp_set_m_type(newaudiortp, global_force_dtmf_relay_pt);
++ codec_pt_order[codec_index++] = global_force_dtmf_relay_pt;
++ ast_rtp_set_rtpmap_type(newaudiortp, global_force_dtmf_relay_pt, "audio", "telephone-event", 0);
++ }
+
+ /* Now gather all of the codecs that we are asked for: */
+ ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
+@@ -16845,6 +16858,9 @@
+
+ global_matchexterniplocally = FALSE;
+
++ global_force_dtmf_relay = 0;
++ global_force_dtmf_relay_pt = 101;
++
+ /* Copy the default jb config over global_jbconf */
+ memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
+
+@@ -16901,6 +16917,18 @@
+ }
+ } else if (!strcasecmp(v->name, "vmexten")) {
+ ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
++ } else if (!strcasecmp(v->name, "rtp_force_dtmf_relay")) {
++ if ((global_force_dtmf_relay = ast_true(v->value)))
++ ast_verbose("RTP DTMF relaying will be enforced\n");
++ else
++ ast_verbose("RTP DTMF relaying will not be enforced\n");
++ } else if (!strcasecmp(v->name, "rtp_force_dtmf_relay_pt")) {
++ sscanf(v->value, "%d", &global_force_dtmf_relay_pt);
++ if (global_force_dtmf_relay_pt < 96 || global_force_dtmf_relay_pt > 255) {
++ ast_verbose("RTP forced DTMF relay payload type is not valid: %d. Using default (101)\n", global_force_dtmf_relay_pt);
++ global_force_dtmf_relay_pt = 101;
++ } else
++ ast_log(LOG_WARNING, "RTP forced DTMF relay payload type is %d\n", global_force_dtmf_relay_pt);
+ } else if (!strcasecmp(v->name, "rtptimeout")) {
+ if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
+ ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
+--- configs/sip.conf.sample.orig 2008-03-12 17:57:19.000000000 +0200
++++ configs/sip.conf.sample 2008-03-12 18:13:03.000000000 +0200
+@@ -53,6 +53,12 @@
+ ; and multiline formatted headers for strict
+ ; SIP compatibility (defaults to "no")
+
++;rtp_force_dtmf_relay=no ; Enable RFC2833 DTMFs to be sent even if peer
++ ; hasn't announced support for it. Default: no
++
++;rtp_force_dtmf_relay_pt=101 ; RTP payload type value for enforced RFC2833
++ ; DTMFs. Default: 101
++
+ ; See doc/README.tos for a description of these parameters.
+ ;tos_sip=cs3 ; Sets TOS for SIP packets.
+ ;tos_audio=ef ; Sets TOS for RTP audio packets.
diff --git a/net/asterisk16/files/rtp_force_dtmf-nocodecnego.diff b/net/asterisk16/files/rtp_force_dtmf-nocodecnego.diff
new file mode 100644
index 000000000000..776422b34645
--- /dev/null
+++ b/net/asterisk16/files/rtp_force_dtmf-nocodecnego.diff
@@ -0,0 +1,70 @@
+--- channels/chan_sip.c.orig 2008-01-31 21:52:49.000000000 +0200
++++ channels/chan_sip.c 2008-03-14 17:50:57.000000000 +0200
+@@ -556,6 +556,9 @@
+ static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
+ static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
+ static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
++static int global_force_dtmf_relay = 0;
++static int global_force_dtmf_relay_pt = 101;
++
+ static int compactheaders; /*!< send compact sip headers */
+ static int recordhistory; /*!< Record SIP history. Off by default */
+ static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
+@@ -5392,6 +5395,13 @@
+
+ /* Now gather all of the codecs that we are asked for: */
+ ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
++ /* Add telephone-event */
++ if (global_force_dtmf_relay && !(peernoncodeccapability & AST_RTP_DTMF)) {
++ ast_rtp_set_m_type(newaudiortp, global_force_dtmf_relay_pt);
++ found_rtpmap_codecs[last_rtpmap_codec++] = global_force_dtmf_relay_pt;
++ ast_rtp_set_rtpmap_type(newaudiortp, global_force_dtmf_relay_pt, "audio", "telephone-event", 0);
++ peernoncodeccapability |= AST_RTP_DTMF;
++ }
+ ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
+
+ newjointcapability = p->capability & (peercapability | vpeercapability);
+@@ -16833,6 +16843,9 @@
+
+ global_matchexterniplocally = FALSE;
+
++ global_force_dtmf_relay = 0;
++ global_force_dtmf_relay_pt = 101;
++
+ /* Copy the default jb config over global_jbconf */
+ memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
+
+@@ -16889,6 +16902,18 @@
+ }
+ } else if (!strcasecmp(v->name, "vmexten")) {
+ ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
++ } else if (!strcasecmp(v->name, "rtp_force_dtmf_relay")) {
++ if ((global_force_dtmf_relay = ast_true(v->value)))
++ ast_verbose("RTP DTMF relaying will be enforced\n");
++ else
++ ast_verbose("RTP DTMF relaying will not be enforced\n");
++ } else if (!strcasecmp(v->name, "rtp_force_dtmf_relay_pt")) {
++ sscanf(v->value, "%d", &global_force_dtmf_relay_pt);
++ if (global_force_dtmf_relay_pt < 96 || global_force_dtmf_relay_pt > 255) {
++ ast_verbose("RTP forced DTMF relay payload type is not valid: %d. Using default (101)\n", global_force_dtmf_relay_pt);
++ global_force_dtmf_relay_pt = 101;
++ } else
++ ast_log(LOG_WARNING, "RTP forced DTMF relay payload type is %d\n", global_force_dtmf_relay_pt);
+ } else if (!strcasecmp(v->name, "rtptimeout")) {
+ if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
+ ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
+--- configs/sip.conf.sample.orig 2008-03-12 17:57:19.000000000 +0200
++++ configs/sip.conf.sample 2008-03-12 18:13:03.000000000 +0200
+@@ -53,6 +53,12 @@
+ ; and multiline formatted headers for strict
+ ; SIP compatibility (defaults to "no")
+
++;rtp_force_dtmf_relay=no ; Enable RFC2833 DTMFs to be sent even if peer
++ ; hasn't announced support for it. Default: no
++
++;rtp_force_dtmf_relay_pt=101 ; RTP payload type value for enforced RFC2833
++ ; DTMFs. Default: 101
++
+ ; See doc/README.tos for a description of these parameters.
+ ;tos_sip=cs3 ; Sets TOS for SIP packets.
+ ;tos_audio=ef ; Sets TOS for RTP audio packets.
diff --git a/net/asterisk16/files/sip_force_callid.diff b/net/asterisk16/files/sip_force_callid.diff
new file mode 100644
index 000000000000..67a3e0192633
--- /dev/null
+++ b/net/asterisk16/files/sip_force_callid.diff
@@ -0,0 +1,12 @@
+--- channels/chan_sip.c.orig 2008-04-09 11:39:09.000000000 +0300
++++ channels/chan_sip.c 2008-04-09 11:40:28.000000000 +0300
+@@ -2998,6 +2998,9 @@
+ p->t38.state = T38_LOCAL_DIRECT;
+ if (option_debug)
+ ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
++ } else if (!strcasecmp(ast_var_name(current), "SIP_FORCE_CALLID")) {
++
++ ast_string_field_set(p, callid, ast_var_value(current));
+ }
+
+ }
diff --git a/net/asterisk16/pkg-plist b/net/asterisk16/pkg-plist
index 7bcd9f5265f4..85ea7e48e505 100644
--- a/net/asterisk16/pkg-plist
+++ b/net/asterisk16/pkg-plist
@@ -295,7 +295,6 @@ lib/asterisk/modules/codec_adpcm.so
lib/asterisk/modules/codec_alaw.so
lib/asterisk/modules/codec_g726.so
lib/asterisk/modules/codec_gsm.so
-lib/asterisk/modules/codec_ilbc.so
lib/asterisk/modules/codec_lpc10.so
lib/asterisk/modules/codec_speex.so
lib/asterisk/modules/codec_ulaw.so
@@ -380,8 +379,9 @@ share/asterisk/moh/LICENSE-asterisk-moh-freeplay-wav
share/asterisk/moh/fpm-calm-river.wav
share/asterisk/moh/fpm-sunshine.wav
share/asterisk/moh/fpm-world-mix.wav
-share/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.8
-share/asterisk/sounds/CREDITS-asterisk-core-en-1.4.8
+share/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.9
+share/asterisk/sounds/CHANGES-asterisk-core-en-1.4.9
+share/asterisk/sounds/CREDITS-asterisk-core-en-1.4.9
share/asterisk/sounds/agent-alreadyon.gsm
share/asterisk/sounds/agent-incorrect.gsm
share/asterisk/sounds/agent-loggedoff.gsm
@@ -526,15 +526,19 @@ share/asterisk/sounds/digits/thousand.gsm
share/asterisk/sounds/digits/today.gsm
share/asterisk/sounds/digits/tomorrow.gsm
share/asterisk/sounds/digits/yesterday.gsm
+share/asterisk/sounds/dir-first.gsm
+share/asterisk/sounds/dir-firstlast.gsm
share/asterisk/sounds/dir-instr.gsm
share/asterisk/sounds/dir-intro-fn.gsm
share/asterisk/sounds/dir-intro.gsm
+share/asterisk/sounds/dir-last.gsm
share/asterisk/sounds/dir-multi1.gsm
share/asterisk/sounds/dir-multi2.gsm
share/asterisk/sounds/dir-multi3.gsm
share/asterisk/sounds/dir-multi9.gsm
share/asterisk/sounds/dir-nomatch.gsm
share/asterisk/sounds/dir-nomore.gsm
+share/asterisk/sounds/dir-pls-enter.gsm
share/asterisk/sounds/followme/call-from.gsm
share/asterisk/sounds/followme/no-recording.gsm
share/asterisk/sounds/followme/options.gsm
@@ -669,12 +673,20 @@ share/asterisk/sounds/silence/7.gsm
share/asterisk/sounds/silence/8.gsm
share/asterisk/sounds/silence/9.gsm
share/asterisk/sounds/spy-agent.gsm
+share/asterisk/sounds/spy-console.gsm
share/asterisk/sounds/spy-h323.gsm
share/asterisk/sounds/spy-iax.gsm
share/asterisk/sounds/spy-iax2.gsm
+share/asterisk/sounds/spy-jingle.gsm
+share/asterisk/sounds/spy-local.gsm
share/asterisk/sounds/spy-mgcp.gsm
+share/asterisk/sounds/spy-misdn.gsm
+share/asterisk/sounds/spy-mobile.gsm
+share/asterisk/sounds/spy-nbs.gsm
share/asterisk/sounds/spy-sip.gsm
share/asterisk/sounds/spy-skinny.gsm
+share/asterisk/sounds/spy-unistim.gsm
+share/asterisk/sounds/spy-usbradio.gsm
share/asterisk/sounds/spy-zap.gsm
share/asterisk/sounds/ss-noservice.gsm
share/asterisk/sounds/transfer.gsm
@@ -716,6 +728,7 @@ share/asterisk/sounds/vm-incorrect-mailbox.gsm
share/asterisk/sounds/vm-incorrect.gsm
share/asterisk/sounds/vm-instructions.gsm
share/asterisk/sounds/vm-intro.gsm
+share/asterisk/sounds/vm-invalidpassword.gsm
share/asterisk/sounds/vm-isonphone.gsm
share/asterisk/sounds/vm-isunavail.gsm
share/asterisk/sounds/vm-last.gsm
@@ -738,8 +751,10 @@ share/asterisk/sounds/vm-nomore.gsm
share/asterisk/sounds/vm-nonumber.gsm
share/asterisk/sounds/vm-num-i-have.gsm
share/asterisk/sounds/vm-onefor.gsm
+share/asterisk/sounds/vm-onefor-full.gsm
share/asterisk/sounds/vm-options.gsm
share/asterisk/sounds/vm-opts.gsm
+share/asterisk/sounds/vm-opts-full.gsm
share/asterisk/sounds/vm-passchanged.gsm
share/asterisk/sounds/vm-password.gsm
share/asterisk/sounds/vm-press.gsm
@@ -767,6 +782,7 @@ share/asterisk/sounds/vm-tempgreeting2.gsm
share/asterisk/sounds/vm-tempremoved.gsm
share/asterisk/sounds/vm-then-pound.gsm
share/asterisk/sounds/vm-theperson.gsm
+share/asterisk/sounds/vm-tmpexists.gsm
share/asterisk/sounds/vm-tocallback.gsm
share/asterisk/sounds/vm-tocallnum.gsm
share/asterisk/sounds/vm-tocancel.gsm
generated by cgit (git 2.34.1) at 2024-11-08 03:38:17 +0800