Reduce channel snapshot creation and publishing by up to 50%.

This change introduces the ability to stage channel snapshot
creation and publishing by suppressing the implicit creation
and publishing that some functions have. Once all operations
are executed the staging is marked as done and a single snapshot
is created and published.

Review: https://reviewboard.asterisk.org/r/2889/

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400265 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_dial.c b/apps/app_dial.c
index cd675fad96e19fa394d7f10788080d052d2acbdc..2fb7fc9acc52d2ae6224873dfa5d1edd143dfd62 100644 (file)
--- 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 7f54d6cfaecb089a0806805786b773d981067995..f1720ba8cea2563b0af69aa4f10a5d58ce29b450 100644 (file)
--- 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 0f4d01ebd0b58ed331b485eb29e894345fdea084..76571d20989cbafad259f56010f89669e2a0d119 100644 (file)
--- 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 853c5f4e783176d42e5fafb70a2a61200561aae7..65a9758f2632e568d8ef5118b2c3d308c0ccf1a1 100644 (file)
--- 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 fc1b75a53e084fae30e15f1b2dcf66963d206810..a44ca57f46f78a313048732f88812278b5f68686 100644 (file)
--- 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 b880f5d9480305f5fcbbf718906c3cea898268c2..62888615fb44da1178bce0a37964a8b6e247924d 100644 (file)
--- 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 e5fd0250690be3b79c37f20f241eaf3ea366bc83..a8cab623830d9787ec98aad3ee2c32ec14d67a65 100644 (file)
--- 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 8fcdebfad704dff61262a62644259a59f0a726d3..7c91200882cace4b10f17d1a98b2dcecf20c2d86 100644 (file)
--- 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 72118ce1041f60546b076ec27ff7a9cfc40e6d10..e13c66c6feaf15e5c7f95105f29696662c7502a6 100644 (file)
--- 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 42e329e50a161532f766824efd7a3c5a40eb7eb9..fcca25c8d1cbdcb40d2de36bfe56fe5c0ba2203f 100644 (file)
--- 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 2a821dd73f24920fd52d1482e52cce87fdd3e387..497fb6e616350f64756c7e8b821239d9b29911ad 100644 (file)
--- 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 7cf592a261f9f7641a3e131eb249dc776b94b695..496de18667eefdb86bbfca62cb334947728cf22a 100644 (file)
--- 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 adb5cb9c836a68218848bd4a8fc23be47cfbe9af..5238679880b9c88b7bc5556f424985c8f52029c4 100644 (file)
--- 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 9f400772375d313a5256e9bccb33d4aff478171b..5def34c2383950d8e4be29ee6cb8a511265e9f73 100644 (file)
--- 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 e6e11bed6011c61a8c4be9b109122bfbd3da2890..166fda2f7806f7f7bc24d8d7813ad9ae07e27572 100644 (file)
--- 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 24dc53f6bb602239e0d22dda580cbf8cc254e492..e952b93b86583aff611df204583659d3b20583af 100644 (file)
--- 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 c0b596ac97a24686790a428a17a1f186eceadec3..7d9c3efdcf8e7ff18d1a659056face2101ddb120 100644 (file)
--- a/include/asterisk/stasis_channels.h
+++ b/include/asterisk/stasis_channels.h
@@ -314,6 +314,22 @@ void ast_multi_channel_blob_add_channel(struct ast_multi_channel_blob *obj,
 void ast_channel_publish_blob(struct ast_channel *chan, struct stasis_message_type *type,
        struct ast_json *blob);
 
+/*!
+ * \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.

diff --git a/main/bridge.c b/main/bridge.c
index 19350ed845b59966e6682e116e8ec7022bd4c390..a972fe7363148562680cd33709fdbba77da63a55 100644 (file)
--- 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 e8ce5e0c53d12d026c4ca473ffffa3cf5b093513..36528f5df82cfc693b7e6f450c6c7f66bfb06fad 100644 (file)
--- 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 3b720fb9ff72cba3bfaa421469aa39299dd9c4fb..6d796e2b9ded7a905577da7fc5ab4a31fc55908e 100644 (file)
--- 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 2a415401f16a6f3fdf3a001ab6a7568834a19216..6b71c54b9c62b0a94147ecbec6b02b0a6cdfe1da 100644 (file)
--- 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 0fbbf5d96734986abdebf07c969bc04e9ce1b946..e1a3853e0f5932d7f65565d10d0f0f662c644461 100644 (file)
--- 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;