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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

20 files changed, 86 insertions, 6 deletions

diff --git a/channels/chan_alsa.c b/channels/chan_alsa.c
index 60bbacbb8..065eff551 100644
--- a/channels/chan_alsa.c
+++ b/channels/chan_alsa.c
@@ -581,6 +581,7 @@ 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_lock(tmp);
 	ast_channel_stage_snapshot(tmp);
 
 	ast_channel_tech_set(tmp, &alsa_tech);
@@ -601,6 +602,7 @@ static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state, const ch
 	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)) {
diff --git a/channels/chan_console.c b/channels/chan_console.c
index 922c53b42..6f694c2c9 100644
--- a/channels/chan_console.c
+++ b/channels/chan_console.c
@@ -428,6 +428,7 @@ static struct ast_channel *console_new(struct console_pvt *pvt, const char *ext,
 		return NULL;
 	}
 
+	ast_channel_lock(chan);
 	ast_channel_stage_snapshot(chan);
 
 	ast_channel_tech_set(chan, &console_tech);
@@ -444,6 +445,7 @@ 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);
+	ast_channel_unlock(chan);
 
 	if (state != AST_STATE_DOWN) {
 		if (ast_pbx_start(chan)) {
diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index 2eba958a8..843254c56 100644
--- a/channels/chan_dahdi.c
+++ b/channels/chan_dahdi.c
@@ -1667,7 +1667,9 @@ static void publish_dahdichannel(struct ast_channel *chan, int span, const char
 		return;
 	}
 
+	ast_channel_lock(chan);
 	ast_channel_publish_blob(chan, dahdichannel_type(), blob);
+	ast_channel_unlock(chan);
 }
 
 /*!
@@ -8916,6 +8918,7 @@ static struct ast_channel *dahdi_new(struct dahdi_pvt *i, int state, int startpb
 		return NULL;
 	}
 
+	ast_channel_lock(tmp);
 	ast_channel_stage_snapshot(tmp);
 
 	if (callid) {
@@ -9096,6 +9099,7 @@ static struct ast_channel *dahdi_new(struct dahdi_pvt *i, int state, int startpb
 		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 @@ static void *analog_ss_thread(void *data)
 						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 @@ static void *analog_ss_thread(void *data)
 								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 @@ static void *analog_ss_thread(void *data)
 
 		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) {
diff --git a/channels/chan_gtalk.c b/channels/chan_gtalk.c
index d6a094941..72c01ac8d 100644
--- a/channels/chan_gtalk.c
+++ b/channels/chan_gtalk.c
@@ -1150,6 +1150,7 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
 		return NULL;
 	}
 
+	ast_channel_lock(tmp);
 	ast_channel_stage_snapshot(tmp);
 
 	ast_channel_tech_set(tmp, &gtalk_tech);
@@ -1226,6 +1227,7 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
 		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 @@ static int gtalk_newcall(struct gtalk *client, ikspak *pak)
 	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),
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 22dba3c18..7ce098415 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -1061,6 +1061,9 @@ static struct ast_channel *__oh323_new(struct oh323_pvt *pvt, int state, const c
 	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 @@ static struct ast_channel *__oh323_new(struct oh323_pvt *pvt, int state, const c
 		}
 		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));
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index e27cc14c6..80081174e 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -5689,11 +5689,15 @@ static struct ast_channel *ast_iax2_new(int callno, int state, iax2_format capab
 	/* 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 @@ static struct ast_channel *ast_iax2_new(int callno, int state, iax2_format capab
 	}
 
 	ast_channel_stage_snapshot_done(tmp);
+	ast_channel_unlock(tmp);
 
 	if (state != AST_STATE_DOWN) {
 		if (ast_pbx_start(tmp)) {
@@ -12234,7 +12239,9 @@ static struct ast_channel *iax2_request(const char *type, struct ast_format_cap
 	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 */
diff --git a/channels/chan_jingle.c b/channels/chan_jingle.c
index 56c15b193..8604d4eb9 100644
--- a/channels/chan_jingle.c
+++ b/channels/chan_jingle.c
@@ -864,6 +864,7 @@ static struct ast_channel *jingle_new(struct jingle *client, struct jingle_pvt *
 		return NULL;
 	}
 
+	ast_channel_lock(tmp);
 	ast_channel_stage_snapshot(tmp);
 
 	ast_channel_tech_set(tmp, &jingle_tech);
@@ -941,6 +942,7 @@ static struct ast_channel *jingle_new(struct jingle *client, struct jingle_pvt *
 		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 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
 	}
 
 	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) {
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 84fb3b225..c10697158 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -1507,6 +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_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 @@ static struct ast_channel *mgcp_new(struct mgcp_subchannel *sub, int state, cons
 		}
 
 		ast_channel_stage_snapshot_done(tmp);
+		ast_channel_unlock(tmp);
 
 		if (state != AST_STATE_DOWN) {
 			if (ast_pbx_start(tmp)) {
@@ -3046,6 +3048,7 @@ static void *mgcp_ss(void *data)
 				} 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 @@ static void *mgcp_ss(void *data)
 						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);
diff --git a/channels/chan_misdn.c b/channels/chan_misdn.c
index 38197c638..a1b9105e6 100644
--- a/channels/chan_misdn.c
+++ b/channels/chan_misdn.c
@@ -5954,7 +5954,9 @@ static int read_config(struct chan_list *ch)
 	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 @@ static int read_config(struct chan_list *ch)
 
 	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 @@ static int read_config(struct chan_list *ch)
 	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 @@ static int read_config(struct chan_list *ch)
 		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 @@ static int read_config(struct chan_list *ch)
 		/* 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 @@ cb_events(enum event_e event, struct misdn_bchannel *bc, void *user_data)
 
 		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 @@ cb_events(enum event_e event, struct misdn_bchannel *bc, void *user_data)
 		}
 
 		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");
 
diff --git a/channels/chan_motif.c b/channels/chan_motif.c
index 1e7ee3cce..85234ad3a 100644
--- a/channels/chan_motif.c
+++ b/channels/chan_motif.c
@@ -785,6 +785,7 @@ static struct ast_channel *jingle_new(struct jingle_endpoint *endpoint, struct j
 	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 @@ static struct ast_channel *jingle_new(struct jingle_endpoint *endpoint, struct j
 	ao2_unlock(endpoint);
 
 	ast_channel_stage_snapshot_done(chan);
+	ast_channel_unlock(chan);
 
 	return chan;
 }
@@ -2412,7 +2414,9 @@ static void jingle_action_session_initiate(struct jingle_endpoint *endpoint, str
 
 	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) {
diff --git a/channels/chan_nbs.c b/channels/chan_nbs.c
index 1d8c59d4e..41f696207 100644
--- a/channels/chan_nbs.c
+++ b/channels/chan_nbs.c
@@ -223,6 +223,7 @@ static struct ast_channel *nbs_new(struct nbs_pvt *i, int state, const char *lin
 	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 @@ static struct ast_channel *nbs_new(struct nbs_pvt *i, int state, const char *lin
 		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));
diff --git a/channels/chan_oss.c b/channels/chan_oss.c
index 5ae18f362..92c7374db 100644
--- a/channels/chan_oss.c
+++ b/channels/chan_oss.c
@@ -799,6 +799,7 @@ static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx,
 	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 @@ static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx,
 	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));
diff --git a/channels/chan_phone.c b/channels/chan_phone.c
index c1a24b8b1..ca69a2f69 100644
--- a/channels/chan_phone.c
+++ b/channels/chan_phone.c
@@ -862,6 +862,7 @@ static struct ast_channel *phone_new(struct phone_pvt *i, int state, char *cntx,
 	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 @@ static struct ast_channel *phone_new(struct phone_pvt *i, int state, char *cntx,
 			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);
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 80cf270af..12fb45969 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -577,13 +577,15 @@ 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 */
 	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 @@ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int s
 		ast_channel_zone_set(chan, zone);
 	}
 
-	ast_endpoint_add_channel(session->endpoint->persistent, chan);
-
 	ast_channel_stage_snapshot_done(chan);
+	ast_channel_unlock(chan);
+
+	ast_endpoint_add_channel(session->endpoint->persistent, chan);
 
 	return chan;
 }
@@ -2030,9 +2033,11 @@ static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct
 	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);
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index c56f7c4c9..fafe71257 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -8106,6 +8106,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 		}
 	}
 
+	ast_channel_lock(tmp);
 	ast_channel_stage_snapshot(tmp);
 
 	/* If we sent in a callid, bind it to the channel. */
@@ -8113,7 +8114,6 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 		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);
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index a65862ff3..191c3cc24 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -4845,6 +4845,7 @@ static void *skinny_newcall(void *data)
 	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 @@ static void *skinny_newcall(void *data)
 	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 @@ 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_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 @@ static struct ast_channel *skinny_new(struct skinny_line *l, struct skinny_subli
 			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)) {
diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index 547cd7c2f..93fd9c502 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -2517,10 +2517,12 @@ static void *unistim_ss(void *data)
 	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 @@ static struct ast_channel *unistim_new(struct unistim_subchannel *sub, int state
 		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 @@ static struct ast_channel *unistim_new(struct unistim_subchannel *sub, int state
 	ast_channel_priority_set(tmp, 1);
 
 	ast_channel_stage_snapshot_done(tmp);
+	ast_channel_unlock(tmp);
 
 	if (state != AST_STATE_DOWN) {
 		if (unistimdebug) {
diff --git a/channels/chan_vpb.cc b/channels/chan_vpb.cc
index 12f27249b..6ee0e0618 100644
--- a/channels/chan_vpb.cc
+++ b/channels/chan_vpb.cc
@@ -2439,6 +2439,7 @@ static struct ast_channel *vpb_new(struct vpb_pvt *me, enum ast_channel_state st
 
 	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 @@ static struct ast_channel *vpb_new(struct vpb_pvt *me, enum ast_channel_state st
 			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;
 
diff --git a/channels/sig_analog.c b/channels/sig_analog.c
index bbf7a3c8c..51d3f1495 100644
--- a/channels/sig_analog.c
+++ b/channels/sig_analog.c
@@ -2120,6 +2120,7 @@ static void *__analog_ss_thread(void *data)
 						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 @@ static void *__analog_ss_thread(void *data)
 							}
 						}
 						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 @@ static void *__analog_ss_thread(void *data)
 
 		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) {
diff --git a/channels/sig_pri.c b/channels/sig_pri.c
index 5def34c23..48c8c33d7 100644
--- a/channels/sig_pri.c
+++ b/channels/sig_pri.c
@@ -2160,7 +2160,9 @@ static void *pri_ss_thread(void *data)
 #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");