[asterisk-commits] mmichelson: trunk r403314 - in /trunk: ./ addons/ apps/ channels/ funcs/ incl...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Dec 3 11:07:34 CST 2013
Author: mmichelson
Date: Tue Dec 3 11:07:29 2013
New Revision: 403314
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=403314
Log:
Add channel locking for channel snapshot creation.
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
........
Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12
Modified:
trunk/ (props changed)
trunk/addons/chan_mobile.c
trunk/addons/chan_ooh323.c
trunk/apps/app_agent_pool.c
trunk/apps/app_confbridge.c
trunk/apps/app_dial.c
trunk/apps/app_disa.c
trunk/apps/app_meetme.c
trunk/apps/app_queue.c
trunk/apps/app_userevent.c
trunk/apps/app_voicemail.c
trunk/channels/chan_alsa.c
trunk/channels/chan_console.c
trunk/channels/chan_dahdi.c
trunk/channels/chan_gtalk.c
trunk/channels/chan_h323.c
trunk/channels/chan_iax2.c
trunk/channels/chan_jingle.c
trunk/channels/chan_mgcp.c
trunk/channels/chan_misdn.c
trunk/channels/chan_motif.c
trunk/channels/chan_nbs.c
trunk/channels/chan_oss.c
trunk/channels/chan_phone.c
trunk/channels/chan_pjsip.c
trunk/channels/chan_sip.c
trunk/channels/chan_skinny.c
trunk/channels/chan_unistim.c
trunk/channels/chan_vpb.cc
trunk/channels/sig_analog.c
trunk/channels/sig_pri.c
trunk/funcs/func_timeout.c
trunk/include/asterisk/aoc.h
trunk/include/asterisk/channel.h
trunk/include/asterisk/channelstate.h
trunk/include/asterisk/stasis_bridges.h
trunk/include/asterisk/stasis_channels.h
trunk/main/bridge.c
trunk/main/bridge_channel.c
trunk/main/cel.c
trunk/main/channel.c
trunk/main/core_local.c
trunk/main/core_unreal.c
trunk/main/dial.c
trunk/main/endpoints.c
trunk/main/pbx.c
trunk/main/pickup.c
trunk/main/stasis_bridges.c
trunk/main/stasis_channels.c
trunk/pbx/pbx_realtime.c
trunk/res/parking/parking_bridge_features.c
trunk/res/parking/parking_manager.c
trunk/res/res_agi.c
trunk/res/res_pjsip_refer.c
trunk/res/res_stasis.c
trunk/tests/test_cdr.c
trunk/tests/test_cel.c
trunk/tests/test_stasis_channels.c
Propchange: trunk/
------------------------------------------------------------------------------
--- branch-12-merged (original)
+++ branch-12-merged Tue Dec 3 11:07:29 2013
@@ -1,1 +1,1 @@
-/branches/12:1-398558,398560-398577,398579-399305,399307-401390,401392-403175,403179,403207,403209,403221,403223,403240,403256,403258,403271,403290,403312
+/branches/12:1-398558,398560-398577,398579-399305,399307-401390,401392-403175,403179,403207,403209,403221,403223,403240,403256,403258,403271,403290,403311-403312
Modified: trunk/addons/chan_mobile.c
URL: http://svnview.digium.com/svn/asterisk/trunk/addons/chan_mobile.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/addons/chan_mobile.c (original)
+++ trunk/addons/chan_mobile.c Tue Dec 3 11:07:29 2013
@@ -861,6 +861,7 @@
goto e_return;
}
+ ast_channel_lock(chn);
ast_channel_tech_set(chn, &mbl_tech);
ast_format_cap_add(ast_channel_nativeformats(chn), &prefformat);
ast_format_copy(ast_channel_rawreadformat(chn), &prefformat);
@@ -878,6 +879,7 @@
if (pvt->sco_socket != -1) {
ast_channel_set_fd(chn, 0, pvt->sco_socket);
}
+ ast_channel_unlock(chn);
return chn;
Modified: trunk/addons/chan_ooh323.c
URL: http://svnview.digium.com/svn/asterisk/trunk/addons/chan_ooh323.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/addons/chan_ooh323.c (original)
+++ trunk/addons/chan_ooh323.c Tue Dec 3 11:07:29 2013
@@ -2153,8 +2153,8 @@
}
ast_queue_control(c, AST_CONTROL_ANSWER);
- ast_channel_unlock(p->owner);
ast_publish_channel_state(c);
+ ast_channel_unlock(p->owner);
}
ast_mutex_unlock(&p->lock);
Modified: trunk/apps/app_agent_pool.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_agent_pool.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/apps/app_agent_pool.c (original)
+++ trunk/apps/app_agent_pool.c Tue Dec 3 11:07:29 2013
@@ -1464,7 +1464,9 @@
ast_bridge_destroy(caller_bridge, AST_CAUSE_USER_BUSY);
}
+ ast_channel_lock(logged);
send_agent_logoff(logged, agent->username, time_logged_in);
+ ast_channel_unlock(logged);
ast_verb(2, "Agent '%s' logged out. Logged in for %ld seconds.\n",
agent->username, time_logged_in);
ast_channel_unref(logged);
@@ -2045,7 +2047,9 @@
ast_verb(2, "Agent '%s' logged in (format %s/%s)\n", agent->username,
ast_getformatname(ast_channel_readformat(chan)),
ast_getformatname(ast_channel_writeformat(chan)));
+ ast_channel_lock(chan);
send_agent_login(chan, agent->username);
+ ast_channel_unlock(chan);
agent_run(agent, chan);
return -1;
Modified: trunk/apps/app_confbridge.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_confbridge.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/apps/app_confbridge.c (original)
+++ trunk/apps/app_confbridge.c Tue Dec 3 11:07:29 2013
@@ -1368,7 +1368,9 @@
}
/* To make sure playback_chan has the same language of that profile */
+ ast_channel_lock(conference->playback_chan);
ast_channel_language_set(conference->playback_chan, conference->b_profile.language);
+ ast_channel_unlock(conference->playback_chan);
ast_debug(1, "Created announcer channel '%s' to conference bridge '%s'\n",
ast_channel_name(conference->playback_chan), conference->name);
Modified: trunk/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_dial.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/apps/app_dial.c (original)
+++ trunk/apps/app_dial.c Tue Dec 3 11:07:29 2013
@@ -2104,6 +2104,7 @@
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_lock(chan);
ast_channel_stage_snapshot(chan);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
@@ -2111,6 +2112,7 @@
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
ast_channel_stage_snapshot_done(chan);
+ ast_channel_unlock(chan);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
@@ -2443,15 +2445,17 @@
continue;
}
+ ast_channel_lock(tc);
ast_channel_stage_snapshot(tc);
+ ast_channel_unlock(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);
}
+
+ ast_channel_lock_both(tc, chan);
pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
-
- ast_channel_lock_both(tc, chan);
/* Setup outgoing SDP to match incoming one */
if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
@@ -2724,8 +2728,10 @@
number = ast_strdupa(number);
}
ast_channel_unlock(peer);
+ ast_channel_lock(chan);
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
ast_channel_stage_snapshot_done(chan);
+ ast_channel_unlock(chan);
if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
@@ -2811,16 +2817,18 @@
/* 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]);
ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
/* peer goes to the same context and extension as chan, so just copy info from chan*/
+ ast_channel_lock(peer);
+ ast_channel_stage_snapshot(peer);
+ ast_clear_flag(ast_channel_flags(peer), AST_FLAG_OUTGOING);
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);
+ ast_channel_unlock(peer);
if (ast_pbx_start(peer)) {
ast_autoservice_chan_hangup_peer(chan, peer);
}
@@ -2970,7 +2978,9 @@
if (!res) {
if (!ast_tvzero(calldurationlimit)) {
struct timeval whentohangup = ast_tvadd(ast_tvnow(), calldurationlimit);
+ ast_channel_lock(peer);
ast_channel_whentohangup_set(peer, &whentohangup);
+ ast_channel_unlock(peer);
}
if (!ast_strlen_zero(dtmfcalled)) {
ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
Modified: trunk/apps/app_disa.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_disa.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/apps/app_disa.c (original)
+++ trunk/apps/app_disa.c Tue Dec 3 11:07:29 2013
@@ -381,8 +381,11 @@
ast_set_callerid(chan, ourcidnum, ourcidname, ourcidnum);
}
- if (!ast_strlen_zero(acctcode))
+ if (!ast_strlen_zero(acctcode)) {
+ ast_channel_lock(chan);
ast_channel_accountcode_set(chan, acctcode);
+ ast_channel_unlock(chan);
+ }
if (special_noanswer) {
ast_clear_flag(&cdr_flags, AST_CDR_FLAG_DISABLE);
Modified: trunk/apps/app_meetme.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_meetme.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/apps/app_meetme.c (original)
+++ trunk/apps/app_meetme.c Tue Dec 3 11:07:29 2013
@@ -1384,7 +1384,9 @@
}
}
+ ast_channel_lock(chan);
msg = ast_channel_blob_create(chan, message_type, json_object);
+ ast_channel_unlock(chan);
if (!msg) {
return;
Modified: trunk/apps/app_queue.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_queue.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/apps/app_queue.c (original)
+++ trunk/apps/app_queue.c Tue Dec 3 11:07:29 2013
@@ -2040,8 +2040,12 @@
RAII_VAR(struct ast_channel_snapshot *, caller_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, agent_snapshot, NULL, ao2_cleanup);
+ ast_channel_lock(caller);
caller_snapshot = ast_channel_snapshot_create(caller);
+ ast_channel_unlock(caller);
+ ast_channel_lock(agent);
agent_snapshot = ast_channel_snapshot_create(agent);
+ ast_channel_unlock(agent);
if (!caller_snapshot || !agent_snapshot) {
return;
@@ -3452,7 +3456,9 @@
"Queue", q->name,
"Position", qe->pos,
"Count", q->count);
+ ast_channel_lock(qe->chan);
ast_channel_publish_blob(qe->chan, queue_caller_join_type(), blob);
+ ast_channel_unlock(qe->chan);
ast_debug(1, "Queue '%s' Join, Channel '%s', Position '%d'\n", q->name, ast_channel_name(qe->chan), qe->pos );
}
ao2_unlock(q);
@@ -3731,7 +3737,9 @@
"Queue", q->name,
"Position", qe->pos,
"Count", q->count);
+ ast_channel_lock(qe->chan);
ast_channel_publish_blob(qe->chan, queue_caller_leave_type(), blob);
+ ast_channel_unlock(qe->chan);
ast_debug(1, "Queue '%s' Leave, Channel '%s'\n", q->name, ast_channel_name(qe->chan));
/* Take us out of the queue */
if (prev) {
@@ -4329,10 +4337,13 @@
"Position", qe->pos,
"OriginalPosition", qe->opos,
"HoldTime", (int)(time(NULL) - qe->start));
- ast_channel_publish_blob(qe->chan, queue_caller_abandon_type(), blob);
qe->parent->callsabandoned++;
ao2_unlock(qe->parent);
+
+ ast_channel_lock(qe->chan);
+ ast_channel_publish_blob(qe->chan, queue_caller_abandon_type(), blob);
+ ast_channel_unlock(qe->chan);
}
/*! \brief RNA == Ring No Answer. Common code that is executed when we try a queue member and they don't answer. */
Modified: trunk/apps/app_userevent.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_userevent.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/apps/app_userevent.c (original)
+++ trunk/apps/app_userevent.c Tue Dec 3 11:07:29 2013
@@ -114,7 +114,9 @@
}
}
+ ast_channel_lock(chan);
ast_channel_publish_blob(chan, ast_channel_user_event_type(), blob);
+ ast_channel_unlock(chan);
return 0;
}
Modified: trunk/apps/app_voicemail.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_voicemail.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/apps/app_voicemail.c (original)
+++ trunk/apps/app_voicemail.c Tue Dec 3 11:07:29 2013
@@ -10958,8 +10958,11 @@
#endif
/* Set language from config to override channel language */
- if (!ast_strlen_zero(vmu->language))
+ if (!ast_strlen_zero(vmu->language)) {
+ ast_channel_lock(chan);
ast_channel_language_set(chan, vmu->language);
+ ast_channel_unlock(chan);
+ }
/* Retrieve urgent, old and new message counts */
ast_debug(1, "Before open_mailbox\n");
Modified: trunk/channels/chan_alsa.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_alsa.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_alsa.c (original)
+++ trunk/channels/chan_alsa.c Tue Dec 3 11:07:29 2013
@@ -581,6 +581,7 @@
if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, linkedid, 0, "ALSA/%s", indevname)))
return NULL;
+ ast_channel_lock(tmp);
ast_channel_stage_snapshot(tmp);
ast_channel_tech_set(tmp, &alsa_tech);
@@ -601,6 +602,7 @@
ast_jb_configure(tmp, &global_jbconf);
ast_channel_stage_snapshot_done(tmp);
+ ast_channel_unlock(tmp);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
Modified: trunk/channels/chan_console.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_console.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_console.c (original)
+++ trunk/channels/chan_console.c Tue Dec 3 11:07:29 2013
@@ -428,6 +428,7 @@
return NULL;
}
+ ast_channel_lock(chan);
ast_channel_stage_snapshot(chan);
ast_channel_tech_set(chan, &console_tech);
@@ -444,6 +445,7 @@
ast_jb_configure(chan, &global_jbconf);
ast_channel_stage_snapshot_done(chan);
+ ast_channel_unlock(chan);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(chan)) {
Modified: trunk/channels/chan_dahdi.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_dahdi.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_dahdi.c (original)
+++ trunk/channels/chan_dahdi.c Tue Dec 3 11:07:29 2013
@@ -1667,7 +1667,9 @@
return;
}
+ ast_channel_lock(chan);
ast_channel_publish_blob(chan, dahdichannel_type(), blob);
+ ast_channel_unlock(chan);
}
/*!
@@ -8916,6 +8918,7 @@
return NULL;
}
+ ast_channel_lock(tmp);
ast_channel_stage_snapshot(tmp);
if (callid) {
@@ -9096,6 +9099,7 @@
pbx_builtin_setvar_helper(tmp, v->name, v->value);
ast_channel_stage_snapshot_done(tmp);
+ ast_channel_unlock(tmp);
ast_module_ref(ast_module_info->self);
@@ -9614,6 +9618,7 @@
getforward = 0;
} else {
res = tone_zone_play_tone(p->subs[idx].dfd, -1);
+ ast_channel_lock(chan);
ast_channel_exten_set(chan, exten);
if (!ast_strlen_zero(p->cid_num)) {
if (!p->hidecallerid)
@@ -9626,6 +9631,7 @@
ast_set_callerid(chan, NULL, p->cid_name, NULL);
}
ast_setstate(chan, AST_STATE_RING);
+ ast_channel_unlock(chan);
dahdi_ec_enable(p);
res = ast_pbx_run(chan);
if (res) {
@@ -10389,8 +10395,10 @@
my_handle_notify_message(chan, p, flags, -1);
+ ast_channel_lock(chan);
ast_setstate(chan, AST_STATE_RING);
ast_channel_rings_set(chan, 1);
+ ast_channel_unlock(chan);
p->ringt = p->ringt_base;
res = ast_pbx_run(chan);
if (res) {
Modified: trunk/channels/chan_gtalk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_gtalk.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_gtalk.c (original)
+++ trunk/channels/chan_gtalk.c Tue Dec 3 11:07:29 2013
@@ -1150,6 +1150,7 @@
return NULL;
}
+ ast_channel_lock(tmp);
ast_channel_stage_snapshot(tmp);
ast_channel_tech_set(tmp, >alk_tech);
@@ -1226,6 +1227,7 @@
ast_jb_configure(tmp, &global_jbconf);
ast_channel_stage_snapshot_done(tmp);
+ ast_channel_unlock(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));
@@ -1419,7 +1421,9 @@
ast_format_cap_joint_copy(p->cap, p->peercap, p->jointcap);
ast_mutex_unlock(&p->lock);
+ ast_channel_lock(chan);
ast_setstate(chan, AST_STATE_RING);
+ ast_channel_unlock(chan);
if (ast_format_cap_is_empty(p->jointcap)) {
ast_log(LOG_WARNING, "Capabilities don't match : us - %s, peer - %s, combined - %s \n", ast_getformatname_multiple(s1, BUFSIZ, p->cap),
ast_getformatname_multiple(s2, BUFSIZ, p->peercap),
Modified: trunk/channels/chan_h323.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_h323.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_h323.c (original)
+++ trunk/channels/chan_h323.c Tue Dec 3 11:07:29 2013
@@ -1061,6 +1061,9 @@
ch = ast_channel_alloc(1, state, cid_num, cid_name, pvt->accountcode, pvt->exten, pvt->context, linkedid, pvt->amaflags, "H323/%s", host);
/* Update usage counter */
ast_module_ref(ast_module_info->self);
+ if (ch) {
+ ast_channel_lock(ch);
+ }
ast_mutex_lock(&pvt->lock);
if (ch) {
ast_channel_tech_set(ch, &oh323_tech);
@@ -1139,6 +1142,7 @@
}
if (pvt->cd.transfer_capability >= 0)
ast_channel_transfercapability_set(ch, pvt->cd.transfer_capability);
+ ast_channel_unlock(ch);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(ch)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(ch));
Modified: trunk/channels/chan_iax2.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_iax2.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_iax2.c (original)
+++ trunk/channels/chan_iax2.c Tue Dec 3 11:07:29 2013
@@ -5689,11 +5689,15 @@
/* Don't hold call lock */
ast_mutex_unlock(&iaxsl[callno]);
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, linkedid, i->amaflags, "IAX2/%s-%d", i->host, i->callno);
+ if (tmp) {
+ ast_channel_lock(tmp);
+ }
ast_mutex_lock(&iaxsl[callno]);
if (i != iaxs[callno]) {
if (tmp) {
/* unlock and relock iaxsl[callno] to preserve locking order */
ast_mutex_unlock(&iaxsl[callno]);
+ ast_channel_unlock(tmp);
tmp = ast_channel_release(tmp);
ast_mutex_lock(&iaxsl[callno]);
}
@@ -5803,6 +5807,7 @@
}
ast_channel_stage_snapshot_done(tmp);
+ ast_channel_unlock(tmp);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
@@ -12234,7 +12239,9 @@
if (c) {
struct ast_format_cap *joint;
if (callid) {
+ ast_channel_lock(c);
ast_channel_callid_set(c, callid);
+ ast_channel_unlock(c);
}
/* Choose a format we can live with */
Modified: trunk/channels/chan_jingle.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_jingle.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_jingle.c (original)
+++ trunk/channels/chan_jingle.c Tue Dec 3 11:07:29 2013
@@ -864,6 +864,7 @@
return NULL;
}
+ ast_channel_lock(tmp);
ast_channel_stage_snapshot(tmp);
ast_channel_tech_set(tmp, &jingle_tech);
@@ -941,6 +942,7 @@
ast_jb_configure(tmp, &global_jbconf);
ast_channel_stage_snapshot_done(tmp);
+ ast_channel_unlock(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));
@@ -1115,7 +1117,9 @@
}
ast_mutex_unlock(&p->lock);
+ ast_channel_lock(chan);
ast_setstate(chan, AST_STATE_RING);
+ ast_channel_unlock(chan);
res = ast_pbx_start(chan);
switch (res) {
Modified: trunk/channels/chan_mgcp.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_mgcp.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_mgcp.c (original)
+++ trunk/channels/chan_mgcp.c Tue Dec 3 11:07:29 2013
@@ -1507,6 +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_lock(tmp);
ast_channel_stage_snapshot(tmp);
ast_channel_tech_set(tmp, &mgcp_tech);
ast_format_cap_copy(ast_channel_nativeformats(tmp), i->cap);
@@ -1570,6 +1571,7 @@
}
ast_channel_stage_snapshot_done(tmp);
+ ast_channel_unlock(tmp);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
@@ -3046,6 +3048,7 @@
} else {
/*res = tone_zone_play_tone(p->subs[index].zfd, -1);*/
ast_indicate(chan, -1);
+ ast_channel_lock(chan);
ast_channel_exten_set(chan, p->dtmf_buf);
ast_channel_dialed(chan)->number.str = ast_strdup(p->dtmf_buf);
memset(p->dtmf_buf, 0, sizeof(p->dtmf_buf));
@@ -3054,6 +3057,7 @@
p->hidecallerid ? "" : p->cid_name,
ast_channel_caller(chan)->ani.number.valid ? NULL : p->cid_num);
ast_setstate(chan, AST_STATE_RING);
+ ast_channel_unlock(chan);
if (p->dtmfmode & MGCP_DTMF_HYBRID) {
p->dtmfmode |= MGCP_DTMF_INBAND;
ast_indicate(chan, -1);
Modified: trunk/channels/chan_misdn.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_misdn.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_misdn.c (original)
+++ trunk/channels/chan_misdn.c Tue Dec 3 11:07:29 2013
@@ -5954,7 +5954,9 @@
chan_misdn_log(1, port, "read_config: Getting Config\n");
misdn_cfg_get(port, MISDN_CFG_LANGUAGE, lang, sizeof(lang));
+ ast_channel_lock(ast);
ast_channel_language_set(ast, lang);
+ ast_channel_unlock(ast);
misdn_cfg_get(port, MISDN_CFG_MUSICCLASS, ch->mohinterpret, sizeof(ch->mohinterpret));
@@ -6000,7 +6002,9 @@
misdn_cfg_get(bc->port, MISDN_CFG_CONTEXT, ch->context, sizeof(ch->context));
+ ast_channel_lock(ast);
ast_channel_context_set(ast, ch->context);
+ ast_channel_unlock(ast);
#ifdef MISDN_1_2
update_pipeline_config(bc);
@@ -6017,8 +6021,10 @@
misdn_cfg_get(port, MISDN_CFG_PICKUPGROUP, &pg, sizeof(pg));
misdn_cfg_get(port, MISDN_CFG_CALLGROUP, &cg, sizeof(cg));
chan_misdn_log(5, port, " --> * CallGrp:%s PickupGrp:%s\n", ast_print_group(buf, sizeof(buf), cg), ast_print_group(buf2, sizeof(buf2), pg));
+ ast_channel_lock(ast);
ast_channel_pickupgroup_set(ast, pg);
ast_channel_callgroup_set(ast, cg);
+ ast_channel_unlock(ast);
misdn_cfg_get(port, MISDN_CFG_NAMEDPICKUPGROUP, &npg, sizeof(npg));
misdn_cfg_get(port, MISDN_CFG_NAMEDCALLGROUP, &ncg, sizeof(ncg));
@@ -6031,8 +6037,10 @@
ast_free(tmp_str);
}
+ ast_channel_lock(ast);
ast_channel_named_pickupgroups_set(ast, npg);
ast_channel_named_callgroups_set(ast, ncg);
+ ast_channel_unlock(ast);
if (ch->originator == ORG_AST) {
char callerid[BUFFERSIZE + 1];
@@ -6086,7 +6094,9 @@
/* Add configured prefix to dialed.number */
misdn_add_number_prefix(bc->port, bc->dialed.number_type, bc->dialed.number, sizeof(bc->dialed.number));
+ ast_channel_lock(ast);
ast_channel_exten_set(ast, bc->dialed.number);
+ ast_channel_unlock(ast);
misdn_cfg_get(bc->port, MISDN_CFG_OVERLAP_DIAL, &ch->overlap_dial, sizeof(ch->overlap_dial));
ast_mutex_init(&ch->overlap_tv_lock);
@@ -10228,8 +10238,10 @@
export_ch(chan, bc, ch);
+ ast_channel_lock(ch->ast);
ast_channel_rings_set(ch->ast, 1);
ast_setstate(ch->ast, AST_STATE_RINGING);
+ ast_channel_unlock(ch->ast);
/* Update asterisk channel caller information */
chan_misdn_log(2, bc->port, " --> TON: %s(%d)\n", misdn_to_str_ton(bc->caller.number_type), bc->caller.number_type);
@@ -10528,7 +10540,9 @@
}
ast_queue_control(ch->ast, AST_CONTROL_RINGING);
+ ast_channel_lock(ch->ast);
ast_setstate(ch->ast, AST_STATE_RINGING);
+ ast_channel_unlock(ch->ast);
cb_log(7, bc->port, " --> Set State Ringing\n");
Modified: trunk/channels/chan_motif.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_motif.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_motif.c (original)
+++ trunk/channels/chan_motif.c Tue Dec 3 11:07:29 2013
@@ -785,6 +785,7 @@
if (!(chan = ast_channel_alloc(1, state, S_OR(title, ""), S_OR(cid_name, ""), "", "", "", linkedid, 0, "Motif/%s-%04lx", str, ast_random() & 0xffff))) {
return NULL;
}
+ ast_channel_lock(chan);
ast_channel_stage_snapshot(chan);
@@ -852,6 +853,7 @@
ao2_unlock(endpoint);
ast_channel_stage_snapshot_done(chan);
+ ast_channel_unlock(chan);
return chan;
}
@@ -2412,7 +2414,9 @@
ao2_link(endpoint->state->sessions, session);
+ ast_channel_lock(chan);
ast_setstate(chan, AST_STATE_RING);
+ ast_channel_unlock(chan);
res = ast_pbx_start(chan);
switch (res) {
Modified: trunk/channels/chan_nbs.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_nbs.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_nbs.c (original)
+++ trunk/channels/chan_nbs.c Tue Dec 3 11:07:29 2013
@@ -223,6 +223,7 @@
struct ast_channel *tmp;
tmp = ast_channel_alloc(1, state, 0, 0, "", "s", context, linkedid, 0, "NBS/%s", i->stream);
if (tmp) {
+ ast_channel_lock(tmp);
ast_channel_tech_set(tmp, &nbs_tech);
ast_channel_set_fd(tmp, 0, nbs_fd(i->nbs));
@@ -239,6 +240,7 @@
ast_channel_language_set(tmp, "");
i->owner = tmp;
i->u = ast_module_user_add(tmp);
+ ast_channel_unlock(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: trunk/channels/chan_oss.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_oss.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_oss.c (original)
+++ trunk/channels/chan_oss.c Tue Dec 3 11:07:29 2013
@@ -799,6 +799,7 @@
c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, linkedid, 0, "Console/%s", o->device + 5);
if (c == NULL)
return NULL;
+ ast_channel_lock(c);
ast_channel_tech_set(c, &oss_tech);
if (o->sounddev < 0)
setformat(o, O_RDWR);
@@ -829,6 +830,7 @@
o->owner = c;
ast_module_ref(ast_module_info->self);
ast_jb_configure(c, &global_jbconf);
+ ast_channel_unlock(c);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(c)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(c));
Modified: trunk/channels/chan_phone.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_phone.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_phone.c (original)
+++ trunk/channels/chan_phone.c Tue Dec 3 11:07:29 2013
@@ -862,6 +862,7 @@
struct ast_format tmpfmt;
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, "", i->ext, i->context, linkedid, 0, "Phone/%s", i->dev + 5);
if (tmp) {
+ ast_channel_lock(tmp);
ast_channel_tech_set(tmp, cur_tech);
ast_channel_set_fd(tmp, 0, i->fd);
/* XXX Switching formats silently causes kernel panics XXX */
@@ -898,6 +899,7 @@
ast_channel_caller(tmp)->ani.number.valid = 1;
ast_channel_caller(tmp)->ani.number.str = ast_strdup(i->cid_num);
}
+ ast_channel_unlock(tmp);
i->owner = tmp;
ast_module_ref(ast_module_info->self);
Modified: trunk/channels/chan_pjsip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_pjsip.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_pjsip.c (original)
+++ trunk/channels/chan_pjsip.c Tue Dec 3 11:07:29 2013
@@ -577,13 +577,15 @@
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 */
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_lock(chan);
+ ast_channel_stage_snapshot(chan);
+
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));
@@ -632,9 +634,10 @@
ast_channel_zone_set(chan, zone);
}
+ ast_channel_stage_snapshot_done(chan);
+ ast_channel_unlock(chan);
+
ast_endpoint_add_channel(session->endpoint->persistent, chan);
-
- ast_channel_stage_snapshot_done(chan);
return chan;
}
@@ -2030,9 +2033,11 @@
switch (status.code) {
case 180:
ast_queue_control(session->channel, AST_CONTROL_RINGING);
+ ast_channel_lock(session->channel);
if (ast_channel_state(session->channel) != AST_STATE_UP) {
ast_setstate(session->channel, AST_STATE_RINGING);
}
+ ast_channel_unlock(session->channel);
break;
case 183:
ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Dec 3 11:07:29 2013
@@ -8106,6 +8106,7 @@
}
}
+ ast_channel_lock(tmp);
ast_channel_stage_snapshot(tmp);
/* If we sent in a callid, bind it to the channel. */
@@ -8113,7 +8114,6 @@
ast_channel_callid_set(tmp, callid);
}
- ast_channel_lock(tmp);
sip_pvt_lock(i);
ast_channel_cc_params_init(tmp, i->cc_params);
ast_channel_caller(tmp)->id.tag = ast_strdup(i->cid_tag);
Modified: trunk/channels/chan_skinny.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_skinny.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_skinny.c (original)
+++ trunk/channels/chan_skinny.c Tue Dec 3 11:07:29 2013
@@ -4845,6 +4845,7 @@
struct skinny_device *d = l->device;
int res = 0;
+ ast_channel_lock(c);
ast_set_callerid(c,
l->hidecallerid ? "" : l->cid_num,
l->hidecallerid ? "" : l->cid_name,
@@ -4858,6 +4859,7 @@
ast_party_name_init(&ast_channel_connected(c)->id.name);
#endif
ast_setstate(c, AST_STATE_RING);
+ ast_channel_unlock(c);
if (!sub->rtp) {
start_rtp(sub);
}
@@ -5424,6 +5426,7 @@
AST_LIST_INSERT_HEAD(&l->sub, sub, list);
//l->activesub = sub;
}
+ ast_channel_lock(tmp);
ast_channel_stage_snapshot(tmp);
ast_channel_tech_set(tmp, &skinny_tech);
ast_channel_tech_pvt_set(tmp, sub);
@@ -5499,6 +5502,7 @@
pbx_builtin_setvar_helper(tmp, v->name, v->value);
ast_channel_stage_snapshot_done(tmp);
+ ast_channel_unlock(tmp);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
Modified: trunk/channels/chan_unistim.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_unistim.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_unistim.c (original)
+++ trunk/channels/chan_unistim.c Tue Dec 3 11:07:29 2013
@@ -2517,10 +2517,12 @@
int res;
ast_verb(3, "Starting switch on '%s@%s-%d' to %s\n", l->name, l->parent->name, sub->softkey, s->device->phone_number);
+ ast_channel_lock(chan);
ast_channel_exten_set(chan, s->device->phone_number);
+ ast_setstate(chan, AST_STATE_RING);
+ ast_channel_unlock(chan);
ast_copy_string(s->device->redial_number, s->device->phone_number,
sizeof(s->device->redial_number));
- ast_setstate(chan, AST_STATE_RING);
res = ast_pbx_run(chan);
if (res) {
ast_log(LOG_WARNING, "PBX exited non-zero\n");
@@ -5563,6 +5565,7 @@
return NULL;
}
+ ast_channel_lock(tmp);
ast_channel_stage_snapshot(tmp);
ast_format_cap_copy(ast_channel_nativeformats(tmp), l->cap);
@@ -5627,6 +5630,7 @@
ast_channel_priority_set(tmp, 1);
ast_channel_stage_snapshot_done(tmp);
+ ast_channel_unlock(tmp);
if (state != AST_STATE_DOWN) {
if (unistimdebug) {
Modified: trunk/channels/chan_vpb.cc
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_vpb.cc?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/chan_vpb.cc (original)
+++ trunk/channels/chan_vpb.cc Tue Dec 3 11:07:29 2013
@@ -2439,6 +2439,7 @@
tmp = ast_channel_alloc(1, state, 0, 0, "", me->ext, me->context, linkedid, AST_AMA_NONE, "%s", me->dev);
if (tmp) {
+ ast_channel_lock(tmp);
if (use_ast_ind == 1){
ast_channel_tech_set(tmp, &vpb_tech_indicate);
} else {
@@ -2471,6 +2472,7 @@
ast_channel_exten_set(tmp, "s");
if (!ast_strlen_zero(me->language))
ast_channel_language_set(tmp, me->language);
+ ast_channel_unlock(tmp);
me->owner = tmp;
Modified: trunk/channels/sig_analog.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sig_analog.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/sig_analog.c (original)
+++ trunk/channels/sig_analog.c Tue Dec 3 11:07:29 2013
@@ -2120,6 +2120,7 @@
getforward = 0;
} else {
res = analog_play_tone(p, idx, -1);
+ ast_channel_lock(chan);
ast_channel_exten_set(chan, exten);
if (!ast_strlen_zero(p->cid_num)) {
if (!p->hidecallerid) {
@@ -2134,6 +2135,7 @@
}
}
ast_setstate(chan, AST_STATE_RING);
+ ast_channel_unlock(chan);
analog_set_echocanceller(p, 1);
res = ast_pbx_run(chan);
if (res) {
@@ -2615,8 +2617,10 @@
analog_handle_notify_message(chan, p, flags, -1);
+ ast_channel_lock(chan);
ast_setstate(chan, AST_STATE_RING);
ast_channel_rings_set(chan, 1);
+ ast_channel_unlock(chan);
analog_set_ringtimeout(p, p->ringt_base);
res = ast_pbx_run(chan);
if (res) {
Modified: trunk/channels/sig_pri.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sig_pri.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/channels/sig_pri.c (original)
+++ trunk/channels/sig_pri.c Tue Dec 3 11:07:29 2013
@@ -2160,7 +2160,9 @@
#endif /* defined(ISSUE_16789) */
sig_pri_set_echocanceller(p, 1);
+ ast_channel_lock(chan);
ast_setstate(chan, AST_STATE_RING);
+ ast_channel_unlock(chan);
res = ast_pbx_run(chan);
if (res) {
ast_log(LOG_WARNING, "PBX exited non-zero!\n");
Modified: trunk/funcs/func_timeout.c
URL: http://svnview.digium.com/svn/asterisk/trunk/funcs/func_timeout.c?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/funcs/func_timeout.c (original)
+++ trunk/funcs/func_timeout.c Tue Dec 3 11:07:29 2013
@@ -155,7 +155,9 @@
switch (*data) {
case 'a':
case 'A':
+ ast_channel_lock(chan);
ast_channel_setwhentohangup_tv(chan, when);
+ ast_channel_unlock(chan);
if (!ast_tvzero(*ast_channel_whentohangup(chan))) {
when = ast_tvadd(when, ast_tvnow());
ast_strftime(timestr, sizeof(timestr), "%Y-%m-%d %H:%M:%S.%3q %Z",
Modified: trunk/include/asterisk/aoc.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/aoc.h?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/include/asterisk/aoc.h (original)
+++ trunk/include/asterisk/aoc.h Tue Dec 3 11:07:29 2013
@@ -497,7 +497,10 @@
*/
int ast_aoc_decoded2str(const struct ast_aoc_decoded *decoded, struct ast_str **msg);
-/*! \brief generate AOC manager event for an AOC-S, AOC-D, or AOC-E msg */
+/*!
+ * \brief generate AOC manager event for an AOC-S, AOC-D, or AOC-E msg
+ * \pre chan is locked
+ */
int ast_aoc_manager_event(const struct ast_aoc_decoded *decoded, struct ast_channel *chan);
/*! \brief get the message type, AOC-D, AOC-E, or AOC Request */
Modified: trunk/include/asterisk/channel.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/channel.h?view=diff&rev=403314&r1=403313&r2=403314
==============================================================================
--- trunk/include/asterisk/channel.h (original)
+++ trunk/include/asterisk/channel.h Tue Dec 3 11:07:29 2013
@@ -1535,8 +1535,7 @@
* \details
* This function sets the absolute time out on a channel (when to hang up).
*
- * \note This function does not require that the channel is locked before
- * calling it.
+ * \pre chan is locked
*
* \return Nothing
* \sa ast_channel_setwhentohangup_tv()
@@ -1552,8 +1551,7 @@
*
* This function sets the absolute time out on a channel (when to hang up).
*
- * \note This function does not require that the channel is locked before
- * calling it.
+ * \pre chan is locked
*
* \return Nothing
* \since 1.6.1
@@ -2336,6 +2334,8 @@
* \brief adds a list of channel variables to a channel
* \param chan the channel
* \param vars a linked list of variables
+ *
+ * \pre chan is locked
*
* \details
* Variable names can be for a regular channel variable or a dialplan function
@@ -3803,6 +3803,15 @@
void ast_channel_##field##_build_va(struct ast_channel *chan, const char *fmt, va_list ap) __attribute__((format(printf, 2, 0))); \
void ast_channel_##field##_build(struct ast_channel *chan, const char *fmt, ...) __attribute__((format(printf, 2, 3)))
+/*!
+ * The following string fields result in channel snapshot creation and
+ * should have the channel locked when called:
+ *
+ * \li language
+ * \li accountcode
+ * \li peeracccount
+ * \li linkedid
+ */
DECLARE_STRINGFIELD_SETTERS_FOR(name);
DECLARE_STRINGFIELD_SETTERS_FOR(language);
DECLARE_STRINGFIELD_SETTERS_FOR(musicclass);
@@ -3854,6 +3863,10 @@
struct timeval ast_channel_sending_dtmf_tv(const struct ast_channel *chan);
void ast_channel_sending_dtmf_tv_set(struct ast_channel *chan, struct timeval value);
enum ama_flags ast_channel_amaflags(const struct ast_channel *chan);
+
+/*!
+ * \pre chan is locked
+ */
[... 1197 lines stripped ...]
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