[svn-commits] file: branch 12 r400265 - in /branches/12: apps/ channels/ include/asterisk/ ...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Oct 2 11:20:37 CDT 2013
Author: file
Date: Wed Oct 2 11:20:25 2013
New Revision: 400265
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=400265
Log:
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/
Modified:
branches/12/apps/app_dial.c
branches/12/channels/chan_alsa.c
branches/12/channels/chan_console.c
branches/12/channels/chan_dahdi.c
branches/12/channels/chan_gtalk.c
branches/12/channels/chan_iax2.c
branches/12/channels/chan_jingle.c
branches/12/channels/chan_mgcp.c
branches/12/channels/chan_motif.c
branches/12/channels/chan_pjsip.c
branches/12/channels/chan_sip.c
branches/12/channels/chan_skinny.c
branches/12/channels/chan_unistim.c
branches/12/channels/sig_pri.c
branches/12/channels/sig_ss7.c
branches/12/include/asterisk/channel.h
branches/12/include/asterisk/stasis_channels.h
branches/12/main/bridge.c
branches/12/main/channel.c
branches/12/main/dial.c
branches/12/main/pbx.c
branches/12/main/stasis_channels.c
Modified: branches/12/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/apps/app_dial.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/apps/app_dial.c (original)
+++ branches/12/apps/app_dial.c Wed Oct 2 11:20:25 2013
@@ -1958,10 +1958,12 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
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 @@
/* 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 @@
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);
}
Modified: branches/12/channels/chan_alsa.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_alsa.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_alsa.c (original)
+++ branches/12/channels/chan_alsa.c Wed Oct 2 11:20:25 2013
@@ -62,6 +62,7 @@
#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 @@
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 @@
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));
Modified: branches/12/channels/chan_console.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_console.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_console.c (original)
+++ branches/12/channels/chan_console.c Wed Oct 2 11:20:25 2013
@@ -76,6 +76,7 @@
#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 @@
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);
@@ -439,6 +442,8 @@
ast_channel_language_set(chan, pvt->language);
ast_jb_configure(chan, &global_jbconf);
+
+ ast_channel_stage_snapshot_done(chan);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(chan)) {
Modified: branches/12/channels/chan_dahdi.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_dahdi.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_dahdi.c (original)
+++ branches/12/channels/chan_dahdi.c Wed Oct 2 11:20:25 2013
@@ -8910,6 +8910,8 @@
return NULL;
}
+ ast_channel_stage_snapshot(tmp);
+
if (callid) {
ast_channel_callid_set(tmp, callid);
}
@@ -9086,6 +9088,8 @@
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);
Modified: branches/12/channels/chan_gtalk.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_gtalk.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_gtalk.c (original)
+++ branches/12/channels/chan_gtalk.c Wed Oct 2 11:20:25 2013
@@ -1149,6 +1149,9 @@
ast_log(LOG_WARNING, "Unable to allocate Gtalk channel structure!\n");
return NULL;
}
+
+ ast_channel_stage_snapshot(tmp);
+
ast_channel_tech_set(tmp, >alk_tech);
/* Select our native format based on codec preference until we receive
@@ -1221,6 +1224,9 @@
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);
Modified: branches/12/channels/chan_iax2.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_iax2.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_iax2.c (original)
+++ branches/12/channels/chan_iax2.c Wed Oct 2 11:20:25 2013
@@ -116,6 +116,7 @@
#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 @@
return NULL;
}
+ ast_channel_stage_snapshot(tmp);
+
if ((callid = iaxs[callno]->callid)) {
ast_channel_callid_set(tmp, callid);
}
@@ -5798,6 +5801,8 @@
}
}
}
+
+ ast_channel_stage_snapshot_done(tmp);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
Modified: branches/12/channels/chan_jingle.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_jingle.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_jingle.c (original)
+++ branches/12/channels/chan_jingle.c Wed Oct 2 11:20:25 2013
@@ -79,6 +79,7 @@
#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 @@
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 @@
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);
Modified: branches/12/channels/chan_mgcp.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_mgcp.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_mgcp.c (original)
+++ branches/12/channels/chan_mgcp.c Wed Oct 2 11:20:25 2013
@@ -85,6 +85,7 @@
#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 @@
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 @@
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));
Modified: branches/12/channels/chan_motif.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_motif.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_motif.c (original)
+++ branches/12/channels/chan_motif.c Wed Oct 2 11:20:25 2013
@@ -76,6 +76,7 @@
#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 @@
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);
@@ -847,6 +850,8 @@
ast_channel_priority_set(chan, 1);
ao2_unlock(endpoint);
+
+ ast_channel_stage_snapshot_done(chan);
return chan;
}
Modified: branches/12/channels/chan_pjsip.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_pjsip.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_pjsip.c (original)
+++ branches/12/channels/chan_pjsip.c Wed Oct 2 11:20:25 2013
@@ -577,6 +577,8 @@
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 */
@@ -631,6 +633,8 @@
}
ast_endpoint_add_channel(session->endpoint->persistent, chan);
+
+ ast_channel_stage_snapshot_done(chan);
return chan;
}
Modified: branches/12/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_sip.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_sip.c (original)
+++ branches/12/channels/chan_sip.c Wed Oct 2 11:20:25 2013
@@ -7285,6 +7285,10 @@
} 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);
}
@@ -7319,6 +7323,9 @@
append_history(p, "RTCPtext", "Quality:%s", quality);
}
pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", quality);
+ }
+ if (p->rtp || p->vrtp || p->trtp) {
+ ast_channel_stage_snapshot_done(oldowner);
}
/* Send a hangup */
@@ -8092,6 +8099,8 @@
}
}
+ ast_channel_stage_snapshot(tmp);
+
/* If we sent in a callid, bind it to the channel. */
if (callid) {
ast_channel_callid_set(tmp, callid);
@@ -8286,6 +8295,8 @@
if (i->do_history) {
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 @@
}
}
+ 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) {
@@ -26593,6 +26608,10 @@
if (p->owner) {
pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", quality);
}
+ }
+
+ 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 */
Modified: branches/12/channels/chan_skinny.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_skinny.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_skinny.c (original)
+++ branches/12/channels/chan_skinny.c Wed Oct 2 11:20:25 2013
@@ -82,6 +82,7 @@
#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 @@
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);
@@ -5495,6 +5497,8 @@
/* Set channel variables for this call from configuration */
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)) {
Modified: branches/12/channels/chan_unistim.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_unistim.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/chan_unistim.c (original)
+++ branches/12/channels/chan_unistim.c Wed Oct 2 11:20:25 2013
@@ -77,7 +77,7 @@
#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 @@
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 @@
}
}
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");
Modified: branches/12/channels/sig_pri.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/sig_pri.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/sig_pri.c (original)
+++ branches/12/channels/sig_pri.c Wed Oct 2 11:20:25 2013
@@ -6473,6 +6473,7 @@
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 @@
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 @@
* 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 @@
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",
Modified: branches/12/channels/sig_ss7.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/sig_ss7.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/channels/sig_ss7.c (original)
+++ branches/12/channels/sig_ss7.c Wed Oct 2 11:20:25 2013
@@ -621,6 +621,8 @@
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
@@ -699,6 +701,8 @@
/* Clear this after we set it */
p->generic_name[0] = 0;
}
+
+ ast_channel_stage_snapshot_done(c);
sig_ss7_unlock_private(p);
ast_channel_unlock(c);
Modified: branches/12/include/asterisk/channel.h
URL: http://svnview.digium.com/svn/asterisk/branches/12/include/asterisk/channel.h?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/include/asterisk/channel.h (original)
+++ branches/12/include/asterisk/channel.h Wed Oct 2 11:20:25 2013
@@ -923,6 +923,11 @@
* 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 */
Modified: branches/12/include/asterisk/stasis_channels.h
URL: http://svnview.digium.com/svn/asterisk/branches/12/include/asterisk/stasis_channels.h?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/include/asterisk/stasis_channels.h (original)
+++ branches/12/include/asterisk/stasis_channels.h Wed Oct 2 11:20:25 2013
@@ -316,6 +316,22 @@
/*!
* \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.
Modified: branches/12/main/bridge.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/main/bridge.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/main/bridge.c (original)
+++ branches/12/main/bridge.c Wed Oct 2 11:20:25 2013
@@ -1151,8 +1151,10 @@
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);
}
/*!
Modified: branches/12/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/main/channel.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/main/channel.c (original)
+++ branches/12/main/channel.c Wed Oct 2 11:20:25 2013
@@ -876,6 +876,9 @@
/* 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 @@
* 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 @@
{
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)
Modified: branches/12/main/dial.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/main/dial.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/main/dial.c (original)
+++ branches/12/main/dial.c Wed Oct 2 11:20:25 2013
@@ -286,6 +286,8 @@
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);
@@ -311,6 +313,8 @@
ast_channel_adsicpe_set(channel->owner, ast_channel_adsicpe(chan));
ast_channel_transfercapability_set(channel->owner, ast_channel_transfercapability(chan));
}
+
+ ast_channel_stage_snapshot_done(channel->owner);
return 0;
}
Modified: branches/12/main/pbx.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/main/pbx.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/main/pbx.c (original)
+++ branches/12/main/pbx.c Wed Oct 2 11:20:25 2013
@@ -10050,7 +10050,9 @@
return -1;
}
- ast_set_variables(dialed, vars);
+ if (vars) {
+ ast_set_variables(dialed, vars);
+ }
if (account) {
ast_channel_accountcode_set(dialed, account);
Modified: branches/12/main/stasis_channels.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/main/stasis_channels.c?view=diff&rev=400265&r1=400264&r2=400265
==============================================================================
--- branches/12/main/stasis_channels.c (original)
+++ branches/12/main/stasis_channels.c Wed Oct 2 11:20:25 2013
@@ -600,10 +600,25 @@
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) {
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