[asterisk-commits] dlee: branch dlee/record r395138 - in /team/dlee/record: ./ apps/ apps/confbr...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jul 23 10:00:15 CDT 2013


Author: dlee
Date: Tue Jul 23 10:00:12 2013
New Revision: 395138

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=395138
Log:
Merged revisions 394776-395136 from http://svn.asterisk.org/svn/asterisk/trunk

Added:
    team/dlee/record/channels/chan_bridge_media.c
      - copied unchanged from r395136, trunk/channels/chan_bridge_media.c
    team/dlee/record/configs/safe_asterisk.conf.sample
      - copied unchanged from r395136, trunk/configs/safe_asterisk.conf.sample
Modified:
    team/dlee/record/   (props changed)
    team/dlee/record/CHANGES
    team/dlee/record/Makefile
    team/dlee/record/apps/app_celgenuserevent.c
    team/dlee/record/apps/app_dial.c
    team/dlee/record/apps/app_directed_pickup.c
    team/dlee/record/apps/app_queue.c
    team/dlee/record/apps/confbridge/conf_chan_announce.c
    team/dlee/record/apps/confbridge/conf_chan_record.c
    team/dlee/record/apps/confbridge/confbridge_manager.c
    team/dlee/record/bridges/bridge_native_rtp.c
    team/dlee/record/channels/chan_dahdi.c
    team/dlee/record/channels/chan_gulp.c
    team/dlee/record/channels/chan_sip.c
    team/dlee/record/channels/iax2/parser.c
    team/dlee/record/channels/sig_analog.c
    team/dlee/record/configs/iax.conf.sample
    team/dlee/record/configs/indications.conf.sample
    team/dlee/record/contrib/realtime/postgresql/realtime.sql
    team/dlee/record/contrib/scripts/safe_asterisk
    team/dlee/record/funcs/func_channel.c
    team/dlee/record/include/asterisk/astobj2.h
    team/dlee/record/include/asterisk/audiohook.h
    team/dlee/record/include/asterisk/cel.h
    team/dlee/record/include/asterisk/channel.h
    team/dlee/record/include/asterisk/core_unreal.h
    team/dlee/record/include/asterisk/logger.h
    team/dlee/record/include/asterisk/res_sip.h
    team/dlee/record/include/asterisk/res_sip_session.h
    team/dlee/record/include/asterisk/stasis_app.h
    team/dlee/record/include/asterisk/stasis_app_playback.h
    team/dlee/record/include/asterisk/stasis_channels.h
    team/dlee/record/include/asterisk/stasis_message_router.h
    team/dlee/record/include/asterisk/stasis_system.h
    team/dlee/record/main/asterisk.c
    team/dlee/record/main/audiohook.c
    team/dlee/record/main/bridging.c
    team/dlee/record/main/ccss.c
    team/dlee/record/main/cdr.c
    team/dlee/record/main/cel.c
    team/dlee/record/main/channel.c
    team/dlee/record/main/core_unreal.c
    team/dlee/record/main/features.c
    team/dlee/record/main/http.c
    team/dlee/record/main/manager.c
    team/dlee/record/main/manager_bridging.c
    team/dlee/record/main/manager_channels.c
    team/dlee/record/main/pbx.c
    team/dlee/record/main/stasis_channels.c
    team/dlee/record/main/stasis_message_router.c
    team/dlee/record/main/stasis_system.c
    team/dlee/record/res/parking/parking_manager.c
    team/dlee/record/res/res_sip.c
    team/dlee/record/res/res_sip/sip_configuration.c
    team/dlee/record/res/res_sip/sip_options.c
    team/dlee/record/res/res_sip_sdp_rtp.c
    team/dlee/record/res/res_sip_session.c
    team/dlee/record/res/res_sip_session.exports.in
    team/dlee/record/res/res_stasis.c
    team/dlee/record/res/res_stasis_http_bridges.c
    team/dlee/record/res/res_stasis_http_channels.c
    team/dlee/record/res/res_stasis_http_playback.c
    team/dlee/record/res/res_stasis_playback.c
    team/dlee/record/res/stasis/app.c
    team/dlee/record/res/stasis/app.h
    team/dlee/record/res/stasis/control.c
    team/dlee/record/res/stasis_http/ari_model_validators.c
    team/dlee/record/res/stasis_http/ari_model_validators.h
    team/dlee/record/res/stasis_http/resource_bridges.c
    team/dlee/record/res/stasis_http/resource_bridges.h
    team/dlee/record/res/stasis_http/resource_channels.c
    team/dlee/record/res/stasis_http/resource_channels.h
    team/dlee/record/rest-api/api-docs/bridges.json
    team/dlee/record/rest-api/api-docs/channels.json
    team/dlee/record/rest-api/api-docs/playback.json
    team/dlee/record/rest-api/api-docs/recordings.json
    team/dlee/record/tests/test_cel.c
    team/dlee/record/tests/test_stasis.c

Propchange: team/dlee/record/
------------------------------------------------------------------------------
--- branch-11-blocked (original)
+++ branch-11-blocked Tue Jul 23 10:00:12 2013
@@ -1,1 +1,1 @@
-/branches/11:373240,375247,375702,385356
+/branches/11:373240,375247,375702,385356,395020

Propchange: team/dlee/record/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.

Propchange: team/dlee/record/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jul 23 10:00:12 2013
@@ -1,1 +1,1 @@
-/trunk:1-394772
+/trunk:1-395137

Modified: team/dlee/record/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/CHANGES?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/CHANGES (original)
+++ team/dlee/record/CHANGES Tue Jul 23 10:00:12 2013
@@ -294,6 +294,10 @@
    the Local channel. This affects the LocalBridge, LocalOptimizationBegin,
    and LocalOptimizationEnd events.
 
+ * The option 'allowmultiplelogin' can now be set or overriden in a particular
+   account. When set in the general context, it will act as the default
+   setting for defined accounts.
+
 AGI (Asterisk Gateway Interface)
 ------------------
  * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
@@ -319,6 +323,27 @@
  * When a CDR is dispatched, user defined CDR variables from both parties are
    included in the resulting CDR. If both parties have the same variable, only
    the Party A value is provided.
+
+CEL (Channel Event Logging)
+------------------
+ * The 'extra' field of all CEL events that use it now consists of a JSON blob
+   with key/value pairs which are defined in the Asterisk 12 CEL documentation.
+
+ * AST_CEL_BLINDTRANSFER events now report the transferee bridge unique
+   identifier, extension, and context in a JSON blob as the extra string
+   instead of the transferee channel name as the peer.
+
+ * AST_CEL_ATTENDEDTRANSFER events now report the peer as NULL and additional
+   information in the 'extra' string as a JSON blob. For transfers that occur
+   between two bridged channels, the 'extra' JSON blob contains the primary
+   bridge unique identifier, the secondary channel name, and the secondary
+   bridge unique identifier. For transfers that occur between a bridged channel
+   and a channel running an app, the 'extra' JSON blob contains the primary
+   bridge unique identifier, the secondary channel name, and the app name.
+
+ * AST_CEL_LOCAL_OPTIMIZE events have been added to convey local channel
+   optimizations with the record occurring for the semi-one channel and
+   the semi-two channel name in the peer field.
 
 Features
 -------------------
@@ -555,6 +580,18 @@
    If no resources exist or all are unavailable the device state is considered
    to be unavailable.
 
+
+Scripts
+------------------
+
+safe_asterisk
+------------------
+ * The safe_asterisk script will now install over previously installations.
+   In previous versions of Asterisk, once installed a 'make install' would
+   skip over safe_asterisk if it was already installed.
+ * Certain options in safe_asterisk can now be configured from the
+   safe_asterisk.conf file. A sample version of this is located in the
+   configs/ folder.
 
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------

Modified: team/dlee/record/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/Makefile?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/Makefile (original)
+++ team/dlee/record/Makefile Tue Jul 23 10:00:12 2013
@@ -558,8 +558,8 @@
 bininstall: _all installdirs $(SUBDIRS_INSTALL) main-bininstall
 	$(INSTALL) -m 755 contrib/scripts/astgenkey "$(DESTDIR)$(ASTSBINDIR)/"
 	$(INSTALL) -m 755 contrib/scripts/autosupport "$(DESTDIR)$(ASTSBINDIR)/"
-	if [ ! -f "$(DESTDIR)$(ASTSBINDIR)/safe_asterisk" -a ! -f /sbin/launchd ]; then \
-		cat contrib/scripts/safe_asterisk | sed 's|__ASTERISK_SBIN_DIR__|$(ASTSBINDIR)|;s|__ASTERISK_VARRUN_DIR__|$(ASTVARRUNDIR)|;s|__ASTERISK_LOG_DIR__|$(ASTLOGDIR)|;' > contrib/scripts/safe.tmp ; \
+	if [ ! -f /sbin/launchd ]; then \
+		cat contrib/scripts/safe_asterisk | sed 's|__ASTERISK_SBIN_DIR__|$(ASTSBINDIR)|;s|__ASTERISK_VARRUN_DIR__|$(ASTVARRUNDIR)|;s|__ASTERISK_LOG_DIR__|$(ASTLOGDIR)|;s|__ASTERISK_ETC_DIR__|$(ASTETCDIR)|;' > contrib/scripts/safe.tmp ; \
 		$(INSTALL) -m 755 contrib/scripts/safe.tmp "$(DESTDIR)$(ASTSBINDIR)/safe_asterisk" ; \
 		rm -f contrib/scripts/safe.tmp ; \
 	fi

Modified: team/dlee/record/apps/app_celgenuserevent.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/apps/app_celgenuserevent.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/apps/app_celgenuserevent.c (original)
+++ team/dlee/record/apps/app_celgenuserevent.c Tue Jul 23 10:00:12 2013
@@ -75,9 +75,9 @@
 	parse = ast_strdupa(data);
 	AST_STANDARD_APP_ARGS(args, parse);
 
-	blob = ast_json_pack("{s: s, s: s}",
+	blob = ast_json_pack("{s: s, s: {s: s}}",
 		"event", args.event,
-		"extra", args.extra);
+		"extra", "extra", args.extra);
 	if (!blob) {
 		return res;
 	}

Modified: team/dlee/record/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/apps/app_dial.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/apps/app_dial.c (original)
+++ team/dlee/record/apps/app_dial.c Tue Jul 23 10:00:12 2013
@@ -60,7 +60,6 @@
 #include "asterisk/stringfields.h"
 #include "asterisk/global_datastores.h"
 #include "asterisk/dsp.h"
-#include "asterisk/cel.h"
 #include "asterisk/aoc.h"
 #include "asterisk/ccss.h"
 #include "asterisk/indications.h"

Modified: team/dlee/record/apps/app_directed_pickup.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/apps/app_directed_pickup.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/apps/app_directed_pickup.c (original)
+++ team/dlee/record/apps/app_directed_pickup.c Tue Jul 23 10:00:12 2013
@@ -46,7 +46,6 @@
 #include "asterisk/features.h"
 #include "asterisk/manager.h"
 #include "asterisk/callerid.h"
-#include "asterisk/cel.h"
 
 #define PICKUPMARK "PICKUPMARK"
 

Modified: team/dlee/record/apps/app_queue.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/apps/app_queue.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/apps/app_queue.c (original)
+++ team/dlee/record/apps/app_queue.c Tue Jul 23 10:00:12 2013
@@ -103,7 +103,6 @@
 #include "asterisk/taskprocessor.h"
 #include "asterisk/aoc.h"
 #include "asterisk/callerid.h"
-#include "asterisk/cel.h"
 #include "asterisk/data.h"
 #include "asterisk/term.h"
 #include "asterisk/dial.h"

Modified: team/dlee/record/apps/confbridge/conf_chan_announce.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/apps/confbridge/conf_chan_announce.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/apps/confbridge/conf_chan_announce.c (original)
+++ team/dlee/record/apps/confbridge/conf_chan_announce.c Tue Jul 23 10:00:12 2013
@@ -134,6 +134,7 @@
 	.send_text = ast_unreal_sendtext,
 	.queryoption = ast_unreal_queryoption,
 	.setoption = ast_unreal_setoption,
+	.properties = AST_CHAN_TP_ANNOUNCER,
 };
 
 struct ast_channel_tech *conf_announce_get_tech(void)

Modified: team/dlee/record/apps/confbridge/conf_chan_record.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/apps/confbridge/conf_chan_record.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/apps/confbridge/conf_chan_record.c (original)
+++ team/dlee/record/apps/confbridge/conf_chan_record.c Tue Jul 23 10:00:12 2013
@@ -86,6 +86,7 @@
 	.call = rec_call,
 	.read = rec_read,
 	.write = rec_write,
+	.properties = AST_CHAN_TP_RECORDER,
 };
 
 struct ast_channel_tech *conf_record_get_tech(void)

Modified: team/dlee/record/apps/confbridge/confbridge_manager.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/apps/confbridge/confbridge_manager.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/apps/confbridge/confbridge_manager.c (original)
+++ team/dlee/record/apps/confbridge/confbridge_manager.c Tue Jul 23 10:00:12 2013
@@ -195,13 +195,16 @@
 {
 	struct ast_bridge_blob *blob = stasis_message_data(message);
 	const char *conference_name;
-	RAII_VAR(struct ast_str *, bridge_text,
-		ast_manager_build_bridge_state_string(blob->bridge, ""),
-		ast_free);
+	RAII_VAR(struct ast_str *, bridge_text, NULL, ast_free);
 	RAII_VAR(struct ast_str *, channel_text, NULL, ast_free);
 
 	ast_assert(blob != NULL);
 	ast_assert(event != NULL);
+
+	bridge_text = ast_manager_build_bridge_state_string(blob->bridge, "");
+	if (!bridge_text) {
+		return;
+	}
 
 	conference_name = ast_json_string_get(ast_json_object_get(blob->blob, "conference"));
 	ast_assert(conference_name != NULL);

Modified: team/dlee/record/bridges/bridge_native_rtp.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/bridges/bridge_native_rtp.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/bridges/bridge_native_rtp.c (original)
+++ team/dlee/record/bridges/bridge_native_rtp.c Tue Jul 23 10:00:12 2013
@@ -45,7 +45,6 @@
 #include "asterisk/bridging_technology.h"
 #include "asterisk/frame.h"
 #include "asterisk/rtp_engine.h"
-#include "asterisk/audiohook.h"
 
 /*! \brief Forward declarations for frame hook usage */
 static int native_rtp_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel);
@@ -85,13 +84,7 @@
 /*! \brief Internal helper function which checks whether the channels are compatible with our native bridging */
 static int native_rtp_bridge_capable(struct ast_channel *chan)
 {
-	if (ast_channel_monitor(chan) || (ast_channel_audiohooks(chan) &&
-		!ast_audiohook_write_list_empty(ast_channel_audiohooks(chan))) ||
-		!ast_framehook_list_contains_no_active(ast_channel_framehooks(chan))) {
-		return 0;
-	} else {
-		return 1;
-	}
+	return ast_channel_has_audio_frame_or_monitor(chan);
 }
 
 /*! \brief Internal helper function which gets all RTP information (glue and instances) relating to the given channels */

Modified: team/dlee/record/channels/chan_dahdi.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/channels/chan_dahdi.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/channels/chan_dahdi.c (original)
+++ team/dlee/record/channels/chan_dahdi.c Tue Jul 23 10:00:12 2013
@@ -107,7 +107,6 @@
 #include "asterisk/callerid.h"
 #include "asterisk/adsi.h"
 #include "asterisk/cli.h"
-#include "asterisk/cel.h"
 #include "asterisk/features.h"
 #include "asterisk/musiconhold.h"
 #include "asterisk/say.h"

Modified: team/dlee/record/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/channels/chan_gulp.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/channels/chan_gulp.c (original)
+++ team/dlee/record/channels/chan_gulp.c Tue Jul 23 10:00:12 2013
@@ -114,7 +114,6 @@
 };
 
 struct gulp_pvt {
-	struct ast_sip_session *session;
 	struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
 };
 
@@ -122,9 +121,6 @@
 {
 	struct gulp_pvt *pvt = obj;
 	int i;
-
-	ao2_cleanup(pvt->session);
-	pvt->session = NULL;
 
 	for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
 		ao2_cleanup(pvt->media[i]);
@@ -336,12 +332,12 @@
 
 static int media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 
 	if (!strcmp(data, "audio")) {
-		return media_offer_read_av(pvt->session, buf, len, AST_FORMAT_TYPE_AUDIO);
+		return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_AUDIO);
 	} else if (!strcmp(data, "video")) {
-		return media_offer_read_av(pvt->session, buf, len, AST_FORMAT_TYPE_VIDEO);
+		return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_VIDEO);
 	}
 
 	return 0;
@@ -349,10 +345,10 @@
 
 static int media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 
 	struct media_offer_data mdata = {
-		.session = pvt->session,
+		.session = channel->session,
 		.value = value
 	};
 
@@ -362,7 +358,7 @@
 		mdata.media_type = AST_FORMAT_TYPE_VIDEO;
 	}
 
-	return ast_sip_push_task_synchronous(pvt->session->serializer, media_offer_write_av, &mdata);
+	return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
 }
 
 static struct ast_custom_function media_offer_function = {
@@ -374,19 +370,24 @@
 /*! \brief Function called by RTP engine to get local audio RTP peer */
 static enum ast_rtp_glue_result gulp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+	struct gulp_pvt *pvt = channel->pvt;
 	struct ast_sip_endpoint *endpoint;
 
-	if (!pvt || !pvt->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
+	if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
 
-	endpoint = pvt->session->endpoint;
+	endpoint = channel->session->endpoint;
 
 	*instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
 	ao2_ref(*instance, +1);
 
 	ast_assert(endpoint != NULL);
+	if (endpoint->media_encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
+		return AST_RTP_GLUE_RESULT_FORBID;
+	}
+
 	if (endpoint->direct_media) {
 		return AST_RTP_GLUE_RESULT_REMOTE;
 	}
@@ -397,24 +398,33 @@
 /*! \brief Function called by RTP engine to get local video RTP peer */
 static enum ast_rtp_glue_result gulp_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
-
-	if (!pvt || !pvt->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+	struct gulp_pvt *pvt = channel->pvt;
+	struct ast_sip_endpoint *endpoint;
+
+	if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
+
+	endpoint = channel->session->endpoint;
 
 	*instance = pvt->media[SIP_MEDIA_VIDEO]->rtp;
 	ao2_ref(*instance, +1);
 
+	ast_assert(endpoint != NULL);
+	if (endpoint->media_encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
+		return AST_RTP_GLUE_RESULT_FORBID;
+	}
+
 	return AST_RTP_GLUE_RESULT_LOCAL;
 }
 
 /*! \brief Function called by RTP engine to get peer capabilities */
 static void gulp_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
-
-	ast_format_cap_copy(result, pvt->session->endpoint->codecs);
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+
+	ast_format_cap_copy(result, channel->session->endpoint->codecs);
 }
 
 static int send_direct_media_request(void *data)
@@ -486,8 +496,9 @@
 		const struct ast_format_cap *cap,
 		int nat_active)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
-	struct ast_sip_session *session = pvt->session;
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+	struct gulp_pvt *pvt = channel->pvt;
+	struct ast_sip_session *session = channel->session;
 	int changed = 0;
 	struct ast_channel *bridge_peer;
 
@@ -544,7 +555,8 @@
 {
 	struct ast_channel *chan;
 	struct ast_format fmt;
-	struct gulp_pvt *pvt;
+	RAII_VAR(struct gulp_pvt *, pvt, NULL, ao2_cleanup);
+	struct ast_sip_channel_pvt *channel;
 
 	if (!(pvt = ao2_alloc(sizeof(*pvt), gulp_pvt_dtor))) {
 		return NULL;
@@ -552,27 +564,28 @@
 
 	if (!(chan = ast_channel_alloc(1, state, S_OR(session->id.number.str, ""), S_OR(session->id.name.str, ""), "", "", "", linkedid, 0, "Gulp/%s-%08x", ast_sorcery_object_get_id(session->endpoint),
 		ast_atomic_fetchadd_int((int *)&chan_idx, +1)))) {
-		ao2_cleanup(pvt);
 		return NULL;
 	}
 
 	ast_channel_tech_set(chan, &gulp_tech);
 
-	ao2_ref(session, +1);
-	pvt->session = session;
+	if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
+		ast_hangup(chan);
+		return NULL;
+	}
+
 	/* If res_sip_session is ever updated to create/destroy ast_sip_session_media
 	 * during a call such as if multiple same-type stream support is introduced,
 	 * these will need to be recaptured as well */
 	pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
 	pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
-	ast_channel_tech_pvt_set(chan, pvt);
+	ast_channel_tech_pvt_set(chan, channel);
 	if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
 		ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(chan));
 	}
 	if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
 		ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(chan));
 	}
-
 
 	if (ast_format_cap_is_empty(session->req_caps) || !ast_format_cap_has_joint(session->req_caps, session->endpoint->codecs)) {
 		ast_format_cap_copy(ast_channel_nativeformats(chan), session->endpoint->codecs);
@@ -637,8 +650,7 @@
 /*! \brief Function called by core when we should answer a Gulp session */
 static int gulp_answer(struct ast_channel *ast)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
-	struct ast_sip_session *session = pvt->session;
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 
 	if (ast_channel_state(ast) == AST_STATE_UP) {
 		return 0;
@@ -646,10 +658,10 @@
 
 	ast_setstate(ast, AST_STATE_UP);
 
-	ao2_ref(session, +1);
-	if (ast_sip_push_task(session->serializer, answer, session)) {
+	ao2_ref(channel->session, +1);
+	if (ast_sip_push_task(channel->session->serializer, answer, channel->session)) {
 		ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
-		ao2_cleanup(session);
+		ao2_cleanup(channel->session);
 		return -1;
 	}
 
@@ -659,8 +671,8 @@
 /*! \brief Function called by core to read any waiting frames */
 static struct ast_frame *gulp_read(struct ast_channel *ast)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
-	struct ast_sip_session *session = pvt->session;
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
+	struct gulp_pvt *pvt = channel->pvt;
 	struct ast_frame *f;
 	struct ast_sip_session_media *media = NULL;
 	int rtcp = 0;
@@ -702,8 +714,8 @@
 		ast_set_write_format(ast, ast_channel_writeformat(ast));
 	}
 
-	if (session->dsp) {
-		f = ast_dsp_process(ast, session->dsp, f);
+	if (channel->session->dsp) {
+		f = ast_dsp_process(ast, channel->session->dsp, f);
 
 		if (f && (f->frametype == AST_FRAME_DTMF)) {
 			ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
@@ -717,7 +729,8 @@
 /*! \brief Function called by core to write frames */
 static int gulp_write(struct ast_channel *ast, struct ast_frame *frame)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
+	struct gulp_pvt *pvt = channel->pvt;
 	struct ast_sip_session_media *media;
 	int res = 0;
 
@@ -764,9 +777,10 @@
 static int fixup(void *data)
 {
 	struct fixup_data *fix_data = data;
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(fix_data->chan);
-
-	fix_data->session->channel = fix_data->chan;
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(fix_data->chan);
+	struct gulp_pvt *pvt = channel->pvt;
+
+	channel->session->channel = fix_data->chan;
 	if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
 		ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(fix_data->chan));
 	}
@@ -780,18 +794,17 @@
 /*! \brief Function called by core to change the underlying owner channel */
 static int gulp_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(newchan);
-	struct ast_sip_session *session = pvt->session;
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
 	struct fixup_data fix_data;
 
-	fix_data.session = session;
+	fix_data.session = channel->session;
 	fix_data.chan = newchan;
 
-	if (session->channel != oldchan) {
-		return -1;
-	}
-
-	if (ast_sip_push_task_synchronous(session->serializer, fixup, &fix_data)) {
+	if (channel->session->channel != oldchan) {
+		return -1;
+	}
+
+	if (ast_sip_push_task_synchronous(channel->session->serializer, fixup, &fix_data)) {
 		ast_log(LOG_WARNING, "Unable to perform channel fixup\n");
 		return -1;
 	}
@@ -990,8 +1003,8 @@
 /*! \brief Function called by core to ask the channel to indicate some sort of condition */
 static int gulp_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
-	struct ast_sip_session *session = pvt->session;
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
+	struct gulp_pvt *pvt = channel->pvt;
 	struct ast_sip_session_media *media;
 	int response_code = 0;
 	int res = 0;
@@ -999,7 +1012,7 @@
 	switch (condition) {
 	case AST_CONTROL_RINGING:
 		if (ast_channel_state(ast) == AST_STATE_RING) {
-			if (session->endpoint->inband_progress) {
+			if (channel->session->endpoint->inband_progress) {
 				response_code = 183;
 				res = -1;
 			} else {
@@ -1008,7 +1021,7 @@
 		} else {
 			res = -1;
 		}
-		ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "Gulp/%s", ast_sorcery_object_get_id(session->endpoint));
+		ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "Gulp/%s", ast_sorcery_object_get_id(channel->session->endpoint));
 		break;
 	case AST_CONTROL_BUSY:
 		if (ast_channel_state(ast) != AST_STATE_UP) {
@@ -1048,19 +1061,19 @@
 	case AST_CONTROL_VIDUPDATE:
 		media = pvt->media[SIP_MEDIA_VIDEO];
 		if (media && media->rtp) {
-			ao2_ref(session, +1);
-
-			if (ast_sip_push_task(session->serializer, transmit_info_with_vidupdate, session)) {
-				ao2_cleanup(session);
+			ao2_ref(channel->session, +1);
+
+			if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
+				ao2_cleanup(channel->session);
 			}
 		} else {
 			res = -1;
 		}
 		break;
 	case AST_CONTROL_CONNECTED_LINE:
-		ao2_ref(session, +1);
-		if (ast_sip_push_task(session->serializer, update_connected_line_information, session)) {
-			ao2_cleanup(session);
+		ao2_ref(channel->session, +1);
+		if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
+			ao2_cleanup(channel->session);
 		}
 		break;
 	case AST_CONTROL_UPDATE_RTP_PEER:
@@ -1095,10 +1108,10 @@
 	}
 
 	if (response_code) {
-		struct indicate_data *ind_data = indicate_data_alloc(session, condition, response_code, data, datalen);
-		if (!ind_data || ast_sip_push_task(session->serializer, indicate, ind_data)) {
+		struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
+		if (!ind_data || ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
 			ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n",
-					response_code, ast_sorcery_object_get_id(session->endpoint));
+					response_code, ast_sorcery_object_get_id(channel->session->endpoint));
 			ao2_cleanup(ind_data);
 			res = -1;
 		}
@@ -1214,15 +1227,14 @@
 /*! \brief Function called by core for Asterisk initiated transfer */
 static int gulp_transfer(struct ast_channel *chan, const char *target)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
-	struct ast_sip_session *session = pvt->session;
-	struct transfer_data *trnf_data = transfer_data_alloc(session, target);
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+	struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
 
 	if (!trnf_data) {
 		return -1;
 	}
 
-	if (ast_sip_push_task(session->serializer, transfer, trnf_data)) {
+	if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
 		ast_log(LOG_WARNING, "Error requesting transfer\n");
 		ao2_cleanup(trnf_data);
 		return -1;
@@ -1234,12 +1246,12 @@
 /*! \brief Function called by core to start a DTMF digit */
 static int gulp_digit_begin(struct ast_channel *chan, char digit)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
-	struct ast_sip_session *session = pvt->session;
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+	struct gulp_pvt *pvt = channel->pvt;
 	struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
 	int res = 0;
 
-	switch (session->endpoint->dtmf) {
+	switch (channel->session->endpoint->dtmf) {
 	case AST_SIP_DTMF_RFC_4733:
 		if (!media || !media->rtp) {
 			return -1;
@@ -1322,21 +1334,21 @@
 /*! \brief Function called by core to stop a DTMF digit */
 static int gulp_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
-	struct ast_sip_session *session = pvt->session;
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
+	struct gulp_pvt *pvt = channel->pvt;
 	struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
 	int res = 0;
 
-	switch (session->endpoint->dtmf) {
+	switch (channel->session->endpoint->dtmf) {
 	case AST_SIP_DTMF_INFO:
 	{
-		struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(session, digit, duration);
+		struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
 
 		if (!dtmf_data) {
 			return -1;
 		}
 
-		if (ast_sip_push_task(session->serializer, transmit_info_dtmf, dtmf_data)) {
+		if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
 			ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
 			ao2_cleanup(dtmf_data);
 			return -1;
@@ -1378,13 +1390,12 @@
 /*! \brief Function called by core to actually start calling a remote party */
 static int gulp_call(struct ast_channel *ast, const char *dest, int timeout)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
-	struct ast_sip_session *session = pvt->session;
-
-	ao2_ref(session, +1);
-	if (ast_sip_push_task(session->serializer, call, session)) {
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
+
+	ao2_ref(channel->session, +1);
+	if (ast_sip_push_task(channel->session->serializer, call, channel->session)) {
 		ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
-		ao2_cleanup(session);
+		ao2_cleanup(channel->session);
 		return -1;
 	}
 
@@ -1484,8 +1495,9 @@
 	pjsip_tx_data *packet = NULL;
 	struct hangup_data *h_data = data;
 	struct ast_channel *ast = h_data->chan;
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
-	struct ast_sip_session *session = pvt->session;
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
+	struct gulp_pvt *pvt = channel->pvt;
+	struct ast_sip_session *session = channel->session;
 	int cause = h_data->cause;
 
 	if (!session->defer_terminate &&
@@ -1507,16 +1519,16 @@
 /*! \brief Function called by core to hang up a Gulp session */
 static int gulp_hangup(struct ast_channel *ast)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
-	struct ast_sip_session *session = pvt->session;
-	int cause = hangup_cause2sip(ast_channel_hangupcause(session->channel));
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
+	struct gulp_pvt *pvt = channel->pvt;
+	int cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
 	struct hangup_data *h_data = hangup_data_alloc(cause, ast);
 
 	if (!h_data) {
 		goto failure;
 	}
 
-	if (ast_sip_push_task(session->serializer, hangup, h_data)) {
+	if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
 		ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
 		goto failure;
 	}
@@ -1527,7 +1539,7 @@
 	/* Go ahead and do our cleanup of the session and channel even if we're not going
 	 * to be able to send our SIP request/response
 	 */
-	clear_session_and_channel(session, ast, pvt);
+	clear_session_and_channel(channel->session, ast, pvt);
 	ao2_cleanup(pvt);
 	ao2_cleanup(h_data);
 
@@ -1665,10 +1677,10 @@
 /*! \brief Function called by core to send text on Gulp session */
 static int gulp_sendtext(struct ast_channel *ast, const char *text)
 {
-	struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
-	struct sendtext_data *data = sendtext_data_create(pvt->session, text);
-
-	if (!data || ast_sip_push_task(pvt->session->serializer, sendtext, data)) {
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
+	struct sendtext_data *data = sendtext_data_create(channel->session, text);
+
+	if (!data || ast_sip_push_task(channel->session->serializer, sendtext, data)) {
 		ao2_ref(data, -1);
 		return -1;
 	}

Modified: team/dlee/record/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/dlee/record/channels/chan_sip.c?view=diff&rev=395138&r1=395137&r2=395138
==============================================================================
--- team/dlee/record/channels/chan_sip.c (original)
+++ team/dlee/record/channels/chan_sip.c Tue Jul 23 10:00:12 2013
@@ -276,7 +276,6 @@
 #include "asterisk/translate.h"
 #include "asterisk/ast_version.h"
 #include "asterisk/event.h"
-#include "asterisk/cel.h"
 #include "asterisk/data.h"
 #include "asterisk/aoc.h"
 #include "asterisk/message.h"
@@ -8044,7 +8043,7 @@
 		return NULL;
 	}
 
-	if (i->relatedpeer) {
+	if (i->relatedpeer && i->relatedpeer->endpoint) {
 		if (ast_endpoint_add_channel(i->relatedpeer->endpoint, tmp)) {
 			ast_channel_unref(tmp);
 			sip_pvt_lock(i);
@@ -10194,6 +10193,7 @@
 				} else if (!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) {
 					secure_audio = 1;
 
+					processed_crypto = 1;
 					if (p->srtp) {
 						ast_set_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK);
 					}
@@ -10276,6 +10276,7 @@
 				} else if (!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) {
 					secure_video = 1;
 
+					processed_crypto = 1;
 					if (p->vsrtp || (p->vsrtp = ast_sdp_srtp_alloc())) {
 						ast_set_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK);
 					}
@@ -13037,13 +13038,17 @@
 static char *crypto_get_attrib(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32)
 {
 	char *a_crypto;
-	char *orig_crypto;
-
-	if (!srtp) {
+	const char *orig_crypto;
+
+	if (!srtp || dtls_enabled) {
 		return NULL;
 	}
 
-	orig_crypto = ast_strdupa(ast_sdp_srtp_get_attrib(srtp, dtls_enabled, default_taglen_32));
+	orig_crypto = ast_sdp_srtp_get_attrib(srtp, dtls_enabled, default_taglen_32);
+	if (ast_strlen_zero(orig_crypto)) {
+		return NULL;
+	}
+
 	if (ast_asprintf(&a_crypto, "a=crypto:%s\r\n", orig_crypto) == -1) {
 		return NULL;
 	}
@@ -15744,7 +15749,6 @@
 static int expire_register(const void *data)
 {
 	struct sip_peer *peer = (struct sip_peer *)data;
-	RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
 
 	if (!peer) {		/* Hmmm. We have no peer. Weird. */
 		return 0;
@@ -15764,11 +15768,14 @@
 		peer->socket.ws_session = NULL;
 	}
 
-	ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_OFFLINE);
-	blob = ast_json_pack("{s: s, s: s}",
-		"peer_status", "Unregistered",
-		"cause", "Expired");
-	ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
+	if (peer->endpoint) {
+		RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
+		ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_OFFLINE);
+		blob = ast_json_pack("{s: s, s: s}",
+			"peer_status", "Unregistered",
+			"cause", "Expired");
+		ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
+	}
 	register_peer_exten(peer, FALSE);	/* Remove regexten */
 	ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);
 
@@ -16013,7 +16020,6 @@
 	int start = 0;
 	int wildcard_found = 0;
 	int single_binding_found = 0;
-	RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
 
 	ast_copy_string(contact, __get_header(req, "Contact", &start), sizeof(contact));
 
@@ -16201,11 +16207,14 @@
 		ast_db_put("SIP/Registry", peer->name, data);
 	}
 
-	ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_ONLINE);
-	blob = ast_json_pack("{s: s, s: s}",
-		"peer_status", "Registered",
-		"address", ast_sockaddr_stringify(&peer->addr));
-	ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
+	if (peer->endpoint) {
+		RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
+		ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_ONLINE);
+		blob = ast_json_pack("{s: s, s: s}",
+			"peer_status", "Registered",
+			"address", ast_sockaddr_stringify(&peer->addr));
+		ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
+	}
 
 	/* Is this a new IP address for us? */
 	if (ast_sockaddr_cmp(&peer->addr, &oldsin)) {
@@ -17209,7 +17218,6 @@
 		/* Create peer if we have autocreate mode enabled */
 		peer = temp_peer(name);
 		if (peer) {
-			RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
 			ao2_t_link(peers, peer, "link peer into peer table");
 			if (!ast_sockaddr_isnull(&peer->addr)) {
 				ao2_t_link(peers_by_ip, peer, "link peer into peers-by-ip table");
@@ -17238,11 +17246,14 @@
 				ast_string_field_set(p, fullcontact, peer->fullcontact);
 				/* Say OK and ask subsystem to retransmit msg counter */
 				transmit_response_with_date(p, "200 OK", req);
-				ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_ONLINE);
-				blob = ast_json_pack("{s: s, s: s}",
-					"peer_status", "Registered",
-					"address", ast_sockaddr_stringify(addr));
-				ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
+				if (peer->endpoint) {
+					RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
+					ast_endpoint_set_state(peer->endpoint, AST_ENDPOINT_ONLINE);
+					blob = ast_json_pack("{s: s, s: s}",
+						"peer_status", "Registered",
+						"address", ast_sockaddr_stringify(addr));
+					ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
+				}
 				send_mwi = 1;
 				res = 0;
 				break;
@@ -17332,7 +17343,9 @@
 			break;
 		}
 
-		ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
+		if (peer->endpoint) {
+			ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob);
+		}
 	}
 	if (peer) {
 		sip_unref_peer(peer, "register_verify: sip_unref_peer: tossing stack peer pointer at end of func");
@@ -23815,7 +23828,6 @@
 	if (statechanged) {
 		const char *s = is_reachable ? "Reachable" : "Lagged";
 		char str_lastms[20];
-		RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
 
 		snprintf(str_lastms, sizeof(str_lastms), "%d", pingtime);
 
@@ -23825,13 +23837,18 @@
 		if (sip_cfg.peer_rtupdate) {

[... 7036 lines stripped ...]



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