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-rw-r--r--apps/app_dial.c13
-rw-r--r--channels/chan_alsa.c6
-rw-r--r--channels/chan_console.c5
-rw-r--r--channels/chan_dahdi.c4
-rw-r--r--channels/chan_gtalk.c6
-rw-r--r--channels/chan_iax2.c5
-rw-r--r--channels/chan_jingle.c7
-rw-r--r--channels/chan_mgcp.c5
-rw-r--r--channels/chan_motif.c5
-rw-r--r--channels/chan_pjsip.c4
-rw-r--r--channels/chan_sip.c19
-rw-r--r--channels/chan_skinny.c4
-rw-r--r--channels/chan_unistim.c7
-rw-r--r--channels/sig_pri.c5
-rw-r--r--channels/sig_ss7.c4
-rw-r--r--include/asterisk/channel.h5
-rw-r--r--include/asterisk/stasis_channels.h16
-rw-r--r--main/bridge.c2
-rw-r--r--main/channel.c9
-rw-r--r--main/dial.c4
-rw-r--r--main/pbx.c4
-rw-r--r--main/stasis_channels.c15
22 files changed, 151 insertions, 3 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c
index cd675fad9..2fb7fc9ac 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -1958,10 +1958,12 @@ static void end_bridge_callback(void *data)
time(&end);
ast_channel_lock(chan);
+ ast_channel_stage_snapshot(chan);
snprintf(buf, sizeof(buf), "%d", ast_channel_get_up_time(chan));
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
snprintf(buf, sizeof(buf), "%d", ast_channel_get_duration(chan));
pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
+ ast_channel_stage_snapshot_done(chan);
ast_channel_unlock(chan);
}
@@ -2096,11 +2098,13 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
struct ast_party_caller caller;
/* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
+ ast_channel_stage_snapshot(chan);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
+ ast_channel_stage_snapshot_done(chan);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
@@ -2431,6 +2435,9 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
chanlist_free(tmp);
continue;
}
+
+ ast_channel_stage_snapshot(tc);
+
ast_channel_get_device_name(tc, device_name, sizeof(device_name));
if (!ignore_cc) {
ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
@@ -2540,6 +2547,8 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
else
ast_channel_exten_set(tc, ast_channel_exten(chan));
+ ast_channel_stage_snapshot_done(tc);
+
ast_channel_unlock(tc);
ast_channel_unlock(chan);
@@ -2690,6 +2699,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
ast_answer(chan);
strcpy(pa.status, "ANSWER");
+ ast_channel_stage_snapshot(chan);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
/* Ah ha! Someone answered within the desired timeframe. Of course after this
we will always return with -1 so that it is hung up properly after the
@@ -2708,6 +2718,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
}
ast_channel_unlock(peer);
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
+ ast_channel_stage_snapshot_done(chan);
if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
@@ -2793,6 +2804,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
/* chan and peer are going into the PBX; as such neither are considered
* outgoing channels any longer */
ast_clear_flag(ast_channel_flags(chan), AST_FLAG_OUTGOING);
+ ast_channel_stage_snapshot(peer);
ast_clear_flag(ast_channel_flags(peer), AST_FLAG_OUTGOING);
ast_replace_subargument_delimiter(opt_args[OPT_ARG_GOTO]);
@@ -2801,6 +2813,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
ast_channel_context_set(peer, ast_channel_context(chan));
ast_channel_exten_set(peer, ast_channel_exten(chan));
ast_channel_priority_set(peer, ast_channel_priority(chan) + 2);
+ ast_channel_stage_snapshot_done(peer);
if (ast_pbx_start(peer)) {
ast_autoservice_chan_hangup_peer(chan, peer);
}
diff --git a/channels/chan_alsa.c b/channels/chan_alsa.c
index 7f54d6cfa..f1720ba8c 100644
--- a/channels/chan_alsa.c
+++ b/channels/chan_alsa.c
@@ -62,6 +62,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/abstract_jb.h"
#include "asterisk/musiconhold.h"
#include "asterisk/poll-compat.h"
+#include "asterisk/stasis_channels.h"
/*! Global jitterbuffer configuration - by default, jb is disabled
* \note Values shown here match the defaults shown in alsa.conf.sample */
@@ -580,6 +581,8 @@ static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state, const ch
if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, linkedid, 0, "ALSA/%s", indevname)))
return NULL;
+ ast_channel_stage_snapshot(tmp);
+
ast_channel_tech_set(tmp, &alsa_tech);
ast_channel_set_fd(tmp, 0, readdev);
ast_format_set(ast_channel_readformat(tmp), AST_FORMAT_SLINEAR, 0);
@@ -596,6 +599,9 @@ static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state, const ch
p->owner = tmp;
ast_module_ref(ast_module_info->self);
ast_jb_configure(tmp, &global_jbconf);
+
+ ast_channel_stage_snapshot_done(tmp);
+
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(tmp));
diff --git a/channels/chan_console.c b/channels/chan_console.c
index 0f4d01ebd..76571d209 100644
--- a/channels/chan_console.c
+++ b/channels/chan_console.c
@@ -76,6 +76,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/musiconhold.h"
#include "asterisk/callerid.h"
#include "asterisk/astobj2.h"
+#include "asterisk/stasis_channels.h"
/*!
* \brief The sample rate to request from PortAudio
@@ -427,6 +428,8 @@ static struct ast_channel *console_new(struct console_pvt *pvt, const char *ext,
return NULL;
}
+ ast_channel_stage_snapshot(chan);
+
ast_channel_tech_set(chan, &console_tech);
ast_format_set(ast_channel_readformat(chan), AST_FORMAT_SLINEAR16, 0);
ast_format_set(ast_channel_writeformat(chan), AST_FORMAT_SLINEAR16, 0);
@@ -440,6 +443,8 @@ static struct ast_channel *console_new(struct console_pvt *pvt, const char *ext,
ast_jb_configure(chan, &global_jbconf);
+ ast_channel_stage_snapshot_done(chan);
+
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(chan)) {
ast_channel_hangupcause_set(chan, AST_CAUSE_SWITCH_CONGESTION);
diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index 853c5f4e7..65a9758f2 100644
--- a/channels/chan_dahdi.c
+++ b/channels/chan_dahdi.c
@@ -8910,6 +8910,8 @@ static struct ast_channel *dahdi_new(struct dahdi_pvt *i, int state, int startpb
return NULL;
}
+ ast_channel_stage_snapshot(tmp);
+
if (callid) {
ast_channel_callid_set(tmp, callid);
}
@@ -9087,6 +9089,8 @@ static struct ast_channel *dahdi_new(struct dahdi_pvt *i, int state, int startpb
for (v = i->vars ; v ; v = v->next)
pbx_builtin_setvar_helper(tmp, v->name, v->value);
+ ast_channel_stage_snapshot_done(tmp);
+
ast_module_ref(ast_module_info->self);
dahdi_ami_channel_event(i, tmp);
diff --git a/channels/chan_gtalk.c b/channels/chan_gtalk.c
index fc1b75a53..a44ca57f4 100644
--- a/channels/chan_gtalk.c
+++ b/channels/chan_gtalk.c
@@ -1149,6 +1149,9 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
ast_log(LOG_WARNING, "Unable to allocate Gtalk channel structure!\n");
return NULL;
}
+
+ ast_channel_stage_snapshot(tmp);
+
ast_channel_tech_set(tmp, &gtalk_tech);
/* Select our native format based on codec preference until we receive
@@ -1221,6 +1224,9 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
ast_channel_priority_set(tmp, 1);
if (i->rtp)
ast_jb_configure(tmp, &global_jbconf);
+
+ ast_channel_stage_snapshot_done(tmp);
+
if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(tmp));
ast_channel_hangupcause_set(tmp, AST_CAUSE_SWITCH_CONGESTION);
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index b880f5d94..62888615f 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -116,6 +116,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/bridge.h"
#include "asterisk/stasis.h"
#include "asterisk/stasis_system.h"
+#include "asterisk/stasis_channels.h"
#include "iax2/include/iax2.h"
#include "iax2/include/firmware.h"
@@ -5702,6 +5703,8 @@ static struct ast_channel *ast_iax2_new(int callno, int state, iax2_format capab
return NULL;
}
+ ast_channel_stage_snapshot(tmp);
+
if ((callid = iaxs[callno]->callid)) {
ast_channel_callid_set(tmp, callid);
}
@@ -5799,6 +5802,8 @@ static struct ast_channel *ast_iax2_new(int callno, int state, iax2_format capab
}
}
+ ast_channel_stage_snapshot_done(tmp);
+
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(tmp));
diff --git a/channels/chan_jingle.c b/channels/chan_jingle.c
index e5fd02506..a8cab6238 100644
--- a/channels/chan_jingle.c
+++ b/channels/chan_jingle.c
@@ -79,6 +79,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/abstract_jb.h"
#include "asterisk/jabber.h"
#include "asterisk/jingle.h"
+#include "asterisk/stasis_channels.h"
#define JINGLE_CONFIG "jingle.conf"
@@ -862,6 +863,9 @@ static struct ast_channel *jingle_new(struct jingle *client, struct jingle_pvt *
ast_log(LOG_WARNING, "Unable to allocate Jingle channel structure!\n");
return NULL;
}
+
+ ast_channel_stage_snapshot(tmp);
+
ast_channel_tech_set(tmp, &jingle_tech);
/* Select our native format based on codec preference until we receive
@@ -935,6 +939,9 @@ static struct ast_channel *jingle_new(struct jingle *client, struct jingle_pvt *
ast_channel_priority_set(tmp, 1);
if (i->rtp)
ast_jb_configure(tmp, &global_jbconf);
+
+ ast_channel_stage_snapshot_done(tmp);
+
if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(tmp));
ast_channel_hangupcause_set(tmp, AST_CAUSE_SWITCH_CONGESTION);
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 8fcdebfad..7c9120088 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -85,6 +85,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/bridge.h"
#include "asterisk/features_config.h"
#include "asterisk/parking.h"
+#include "asterisk/stasis_channels.h"
/*
* Define to work around buggy dlink MGCP phone firmware which
@@ -1506,6 +1507,7 @@ static struct ast_channel *mgcp_new(struct mgcp_subchannel *sub, int state, cons
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, linkedid, i->accountcode, i->exten, i->context, i->amaflags, "MGCP/%s@%s-%d", i->name, i->parent->name, sub->id);
if (tmp) {
+ ast_channel_stage_snapshot(tmp);
ast_channel_tech_set(tmp, &mgcp_tech);
ast_format_cap_copy(ast_channel_nativeformats(tmp), i->cap);
if (ast_format_cap_is_empty(ast_channel_nativeformats(tmp))) {
@@ -1566,6 +1568,9 @@ static struct ast_channel *mgcp_new(struct mgcp_subchannel *sub, int state, cons
if (sub->rtp) {
ast_jb_configure(tmp, &global_jbconf);
}
+
+ ast_channel_stage_snapshot_done(tmp);
+
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(tmp));
diff --git a/channels/chan_motif.c b/channels/chan_motif.c
index 72118ce10..e13c66c6f 100644
--- a/channels/chan_motif.c
+++ b/channels/chan_motif.c
@@ -76,6 +76,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/astobj.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/xmpp.h"
+#include "asterisk/stasis_channels.h"
/*** DOCUMENTATION
<configInfo name="chan_motif" language="en_US">
@@ -785,6 +786,8 @@ static struct ast_channel *jingle_new(struct jingle_endpoint *endpoint, struct j
return NULL;
}
+ ast_channel_stage_snapshot(chan);
+
ast_channel_tech_set(chan, &jingle_tech);
ast_channel_tech_pvt_set(chan, session);
jingle_set_owner(session, chan);
@@ -848,6 +851,8 @@ static struct ast_channel *jingle_new(struct jingle_endpoint *endpoint, struct j
ao2_unlock(endpoint);
+ ast_channel_stage_snapshot_done(chan);
+
return chan;
}
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 42e329e50..fcca25c8d 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -577,6 +577,8 @@ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int s
return NULL;
}
+ ast_channel_stage_snapshot(chan);
+
/* If res_pjsip_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 */
@@ -632,6 +634,8 @@ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int s
ast_endpoint_add_channel(session->endpoint->persistent, chan);
+ ast_channel_stage_snapshot_done(chan);
+
return chan;
}
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2a821dd73..497fb6e61 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7285,6 +7285,10 @@ static int sip_hangup(struct ast_channel *ast)
} while (sip_pvt_trylock(p));
}
+ if (p->rtp || p->vrtp || p->trtp) {
+ ast_channel_stage_snapshot(oldowner);
+ }
+
if (p->rtp) {
ast_rtp_instance_set_stats_vars(oldowner, p->rtp);
}
@@ -7320,6 +7324,9 @@ static int sip_hangup(struct ast_channel *ast)
}
pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", quality);
}
+ if (p->rtp || p->vrtp || p->trtp) {
+ ast_channel_stage_snapshot_done(oldowner);
+ }
/* Send a hangup */
if (ast_channel_state(oldowner) == AST_STATE_UP) {
@@ -8092,6 +8099,8 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
}
}
+ ast_channel_stage_snapshot(tmp);
+
/* If we sent in a callid, bind it to the channel. */
if (callid) {
ast_channel_callid_set(tmp, callid);
@@ -8287,6 +8296,8 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
append_history(i, "NewChan", "Channel %s - from %s", ast_channel_name(tmp), i->callid);
}
+ ast_channel_stage_snapshot_done(tmp);
+
return tmp;
}
@@ -26531,6 +26542,10 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
}
}
+ if ((p->rtp || p->vrtp || p->trtp) && p->owner) {
+ ast_channel_stage_snapshot(p->owner);
+ }
+
/* Get RTCP quality before end of call */
if (p->rtp) {
if (p->do_history) {
@@ -26595,6 +26610,10 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
}
}
+ if ((p->rtp || p->vrtp || p->trtp) && p->owner) {
+ ast_channel_stage_snapshot_done(p->owner);
+ }
+
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
stop_session_timer(p); /* Stop Session-Timer */
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 7cf592a26..496de1866 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -82,6 +82,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/stasis_endpoints.h"
#include "asterisk/bridge.h"
#include "asterisk/parking.h"
+#include "asterisk/stasis_channels.h"
/*** DOCUMENTATION
<manager name="SKINNYdevices" language="en_US">
@@ -5423,6 +5424,7 @@ static struct ast_channel *skinny_new(struct skinny_line *l, struct skinny_subli
AST_LIST_INSERT_HEAD(&l->sub, sub, list);
//l->activesub = sub;
}
+ ast_channel_stage_snapshot(tmp);
ast_channel_tech_set(tmp, &skinny_tech);
ast_channel_tech_pvt_set(tmp, sub);
ast_format_cap_copy(ast_channel_nativeformats(tmp), l->cap);
@@ -5496,6 +5498,8 @@ static struct ast_channel *skinny_new(struct skinny_line *l, struct skinny_subli
for (v = l->chanvars ; v ; v = v->next)
pbx_builtin_setvar_helper(tmp, v->name, v->value);
+ ast_channel_stage_snapshot_done(tmp);
+
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(tmp));
diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index adb5cb9c8..523867988 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -77,7 +77,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/astdb.h"
#include "asterisk/features_config.h"
#include "asterisk/bridge.h"
-
+#include "asterisk/stasis_channels.h"
#define DEFAULTCONTEXT "default"
#define DEFAULTCALLERID "Unknown"
@@ -5563,6 +5563,8 @@ static struct ast_channel *unistim_new(struct unistim_subchannel *sub, int state
return NULL;
}
+ ast_channel_stage_snapshot(tmp);
+
ast_format_cap_copy(ast_channel_nativeformats(tmp), l->cap);
if (ast_format_cap_is_empty(ast_channel_nativeformats(tmp))) {
ast_format_cap_copy(ast_channel_nativeformats(tmp), global_cap);
@@ -5623,6 +5625,9 @@ static struct ast_channel *unistim_new(struct unistim_subchannel *sub, int state
}
}
ast_channel_priority_set(tmp, 1);
+
+ ast_channel_stage_snapshot_done(tmp);
+
if (state != AST_STATE_DOWN) {
if (unistimdebug) {
ast_verb(0, "Starting pbx in unistim_new\n");
diff --git a/channels/sig_pri.c b/channels/sig_pri.c
index 9f4007723..5def34c23 100644
--- a/channels/sig_pri.c
+++ b/channels/sig_pri.c
@@ -6473,6 +6473,7 @@ static void *pri_dchannel(void *vpri)
ast_mutex_lock(&pri->lock);
sig_pri_lock_private(pri->pvts[chanpos]);
if (c) {
+ ast_channel_stage_snapshot(c);
#if defined(HAVE_PRI_SUBADDR)
if (e->ring.calling.subaddress.valid) {
/* Set Calling Subaddress */
@@ -6557,6 +6558,7 @@ static void *pri_dchannel(void *vpri)
PVT_TO_CHANNEL(pri->pvts[chanpos]), 1);
#endif
}
+ ast_channel_stage_snapshot_done(c);
}
if (c && !ast_pthread_create_detached(&threadid, NULL, pri_ss_thread, pri->pvts[chanpos])) {
ast_verb(3, "Accepting overlap call from '%s' to '%s' on channel %d/%d, span %d\n",
@@ -6602,6 +6604,7 @@ static void *pri_dchannel(void *vpri)
* will do anything with the channel we have just
* created.
*/
+ ast_channel_stage_snapshot(c);
#if defined(HAVE_PRI_SUBADDR)
if (e->ring.calling.subaddress.valid) {
/* Set Calling Subaddress */
@@ -6670,6 +6673,8 @@ static void *pri_dchannel(void *vpri)
sig_pri_handle_subcmds(pri, chanpos, e->e, e->ring.subcmds,
e->ring.call);
+
+ ast_channel_stage_snapshot_done(c);
}
if (c && !ast_pbx_start(c)) {
ast_verb(3, "Accepting call from '%s' to '%s' on channel %d/%d, span %d\n",
diff --git a/channels/sig_ss7.c b/channels/sig_ss7.c
index e6e11bed6..166fda2f7 100644
--- a/channels/sig_ss7.c
+++ b/channels/sig_ss7.c
@@ -621,6 +621,8 @@ static void ss7_start_call(struct sig_ss7_chan *p, struct sig_ss7_linkset *links
sig_ss7_set_echocanceller(p, 1);
+ ast_channel_stage_snapshot(c);
+
/*
* It is reasonably safe to set the following
* channel variables while the channel private
@@ -700,6 +702,8 @@ static void ss7_start_call(struct sig_ss7_chan *p, struct sig_ss7_linkset *links
p->generic_name[0] = 0;
}
+ ast_channel_stage_snapshot_done(c);
+
sig_ss7_unlock_private(p);
ast_channel_unlock(c);
diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h
index 24dc53f6b..e952b93b8 100644
--- a/include/asterisk/channel.h
+++ b/include/asterisk/channel.h
@@ -923,6 +923,11 @@ enum {
* occur.
*/
AST_FLAG_DEAD = (1 << 24),
+ /*!
+ * Channel snapshot should not be published, it is being staged for an explicit
+ * publish.
+ */
+ AST_FLAG_SNAPSHOT_STAGE = (1 << 25),
};
/*! \brief ast_bridge_config flags */
diff --git a/include/asterisk/stasis_channels.h b/include/asterisk/stasis_channels.h
index c0b596ac9..7d9c3efdc 100644
--- a/include/asterisk/stasis_channels.h
+++ b/include/asterisk/stasis_channels.h
@@ -316,6 +316,22 @@ void ast_channel_publish_blob(struct ast_channel *chan, struct stasis_message_ty
/*!
* \since 12
+ * \brief Set flag to indicate channel snapshot is being staged.
+ *
+ * \param chan Channel being staged.
+ */
+void ast_channel_stage_snapshot(struct ast_channel *chan);
+
+/*!
+ * \since 12
+ * \brief Clear flag to indicate channel snapshot is being staged, and publish snapshot.
+ *
+ * \param chan Channel being staged.
+ */
+void ast_channel_stage_snapshot_done(struct ast_channel *chan);
+
+/*!
+ * \since 12
* \brief Publish a \ref ast_channel_snapshot for a channel.
*
* \param chan Channel to publish.
diff --git a/main/bridge.c b/main/bridge.c
index 19350ed84..a972fe736 100644
--- a/main/bridge.c
+++ b/main/bridge.c
@@ -1151,8 +1151,10 @@ static void check_bridge_play_sounds(struct ast_bridge *bridge)
static void update_bridge_vars_set(struct ast_channel *chan, const char *name, const char *pvtid)
{
+ ast_channel_stage_snapshot(chan);
pbx_builtin_setvar_helper(chan, "BRIDGEPEER", name);
pbx_builtin_setvar_helper(chan, "BRIDGEPVTCALLID", pvtid);
+ ast_channel_stage_snapshot_done(chan);
}
/*!
diff --git a/main/channel.c b/main/channel.c
index e8ce5e0c5..36528f5df 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -876,6 +876,9 @@ __ast_channel_alloc_ap(int needqueue, int state, const char *cid_num, const char
/* Channel structure allocation failure. */
return NULL;
}
+
+ ast_channel_stage_snapshot(tmp);
+
if (!(nativeformats = ast_format_cap_alloc())) {
ao2_ref(tmp, -1);
/* format capabilities structure allocation failure */
@@ -1020,7 +1023,7 @@ __ast_channel_alloc_ap(int needqueue, int state, const char *cid_num, const char
* And now, since the channel structure is built, and has its name, let
* the world know of its existance
*/
- ast_channel_publish_snapshot(tmp);
+ ast_channel_stage_snapshot_done(tmp);
return tmp;
}
@@ -7716,8 +7719,12 @@ void ast_set_variables(struct ast_channel *chan, struct ast_variable *vars)
{
struct ast_variable *cur;
+ ast_channel_stage_snapshot(chan);
+
for (cur = vars; cur; cur = cur->next)
pbx_builtin_setvar_helper(chan, cur->name, cur->value);
+
+ ast_channel_stage_snapshot_done(chan);
}
static void *silence_generator_alloc(struct ast_channel *chan, void *data)
diff --git a/main/dial.c b/main/dial.c
index 3b720fb9f..6d796e2b9 100644
--- a/main/dial.c
+++ b/main/dial.c
@@ -286,6 +286,8 @@ static int begin_dial_prerun(struct ast_dial_channel *channel, struct ast_channe
cap_request = NULL;
cap_all_audio = ast_format_cap_destroy(cap_all_audio);
+ ast_channel_stage_snapshot(channel->owner);
+
ast_channel_appl_set(channel->owner, "AppDial2");
ast_channel_data_set(channel->owner, "(Outgoing Line)");
ast_publish_channel_state(channel->owner);
@@ -312,6 +314,8 @@ static int begin_dial_prerun(struct ast_dial_channel *channel, struct ast_channe
ast_channel_transfercapability_set(channel->owner, ast_channel_transfercapability(chan));
}
+ ast_channel_stage_snapshot_done(channel->owner);
+
return 0;
}
diff --git a/main/pbx.c b/main/pbx.c
index 2a415401f..6b71c54b9 100644
--- a/main/pbx.c
+++ b/main/pbx.c
@@ -10050,7 +10050,9 @@ static int pbx_outgoing_attempt(const char *type, struct ast_format_cap *cap, co
return -1;
}
- ast_set_variables(dialed, vars);
+ if (vars) {
+ ast_set_variables(dialed, vars);
+ }
if (account) {
ast_channel_accountcode_set(dialed, account);
diff --git a/main/stasis_channels.c b/main/stasis_channels.c
index 0fbbf5d96..e1a3853e0 100644
--- a/main/stasis_channels.c
+++ b/main/stasis_channels.c
@@ -600,11 +600,26 @@ struct ast_json *ast_multi_channel_blob_get_json(struct ast_multi_channel_blob *
return obj->blob;
}
+void ast_channel_stage_snapshot(struct ast_channel *chan)
+{
+ ast_set_flag(ast_channel_flags(chan), AST_FLAG_SNAPSHOT_STAGE);
+}
+
+void ast_channel_stage_snapshot_done(struct ast_channel *chan)
+{
+ ast_clear_flag(ast_channel_flags(chan), AST_FLAG_SNAPSHOT_STAGE);
+ ast_channel_publish_snapshot(chan);
+}
+
void ast_channel_publish_snapshot(struct ast_channel *chan)
{
RAII_VAR(struct ast_channel_snapshot *, snapshot, NULL, ao2_cleanup);
RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
+ if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_SNAPSHOT_STAGE)) {
+ return;
+ }
+
snapshot = ast_channel_snapshot_create(chan);
if (!snapshot) {
return;
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-28 10:38:12 +0000