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https://github.com/asterisk/asterisk.git
synced 2026-07-22 03:47:12 -07:00
Fix some memory leaks.
These memory leaks were found and fixed by John Hardin. I'm just committing them for him. ASTERISK-24736 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4389 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
@@ -2058,7 +2058,7 @@ static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct p
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static int call_pickup_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
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{
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struct ast_features_pickup_config *pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
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struct ast_features_pickup_config *pickup_cfg;
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struct ast_channel *chan;
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/* We don't care about reinvites */
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@@ -2066,6 +2066,7 @@ static int call_pickup_incoming_request(struct ast_sip_session *session, pjsip_r
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return 0;
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}
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pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
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if (!pickup_cfg) {
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ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension.\n");
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return 0;
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@@ -345,6 +345,7 @@ static void after_bridge_goto_destroy(void *data)
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ast_free((char *) after_bridge->parseable_goto);
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ast_free((char *) after_bridge->context);
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ast_free((char *) after_bridge->exten);
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ast_free((char *) after_bridge);
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}
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/*!
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+2
-4
@@ -4952,8 +4952,6 @@ static void *fast_originate(void *data)
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S_OR(in->cid_name, NULL),
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in->vars, in->account, &chan, in->early_media, &assignedids);
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}
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/* Any vars memory was passed to the ast_pbx_outgoing_xxx() calls. */
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in->vars = NULL;
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if (!chan) {
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snprintf(requested_channel, AST_CHANNEL_NAME, "%s/%s", in->tech, in->data);
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@@ -5400,11 +5398,11 @@ static int action_originate(struct mansession *s, const struct message *m)
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}
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} else if (!ast_strlen_zero(app)) {
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res = ast_pbx_outgoing_app(tech, cap, data, to, app, appdata, &reason, 1, l, n, vars, account, NULL, assignedids.uniqueid ? &assignedids : NULL);
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/* Any vars memory was passed to ast_pbx_outgoing_app(). */
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ast_variables_destroy(vars);
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} else {
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if (exten && context && pi) {
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res = ast_pbx_outgoing_exten(tech, cap, data, to, context, exten, pi, &reason, 1, l, n, vars, account, NULL, bridge_early, assignedids.uniqueid ? &assignedids : NULL);
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/* Any vars memory was passed to ast_pbx_outgoing_exten(). */
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ast_variables_destroy(vars);
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} else {
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astman_send_error(s, m, "Originate with 'Exten' requires 'Context' and 'Priority'");
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ast_variables_destroy(vars);
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+1
-1
@@ -8827,7 +8827,7 @@ static void context_merge(struct ast_context **extcontexts, struct ast_hashtab *
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dupdstr = ast_strdup(prio_item->data);
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res1 = ast_add_extension2(new, 0, prio_item->exten, prio_item->priority, prio_item->label,
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prio_item->matchcid ? prio_item->cidmatch : NULL, prio_item->app, dupdstr, prio_item->datad, prio_item->registrar);
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prio_item->matchcid ? prio_item->cidmatch : NULL, prio_item->app, dupdstr, ast_free_ptr, prio_item->registrar);
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if (!res1 && new_exten_item && new_prio_item){
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ast_verb(3,"Dropping old dialplan item %s/%s/%d [%s(%s)] (registrar=%s) due to conflict with new dialplan\n",
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context->name, prio_item->exten, prio_item->priority, prio_item->app, (char*)prio_item->data, prio_item->registrar);
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@@ -1004,13 +1004,14 @@ static struct ast_json *dtmf_end_to_json(
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const char *direction =
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ast_json_string_get(ast_json_object_get(blob, "direction"));
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const struct timeval *tv = stasis_message_timestamp(message);
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struct ast_json *json_channel = ast_channel_snapshot_to_json(snapshot, sanitize);
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struct ast_json *json_channel;
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/* Only present received DTMF end events as JSON */
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if (strcasecmp("Received", direction) != 0) {
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return NULL;
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}
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json_channel = ast_channel_snapshot_to_json(snapshot, sanitize);
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if (!json_channel) {
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return NULL;
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}
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+10
-3
@@ -2533,7 +2533,8 @@ static struct ast_xml_doc_item *xmldoc_build_list_responses(struct ast_xml_node
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/* Iterate over managerEvent nodes */
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for (event = ast_xml_node_get_children(list_elements); event; event = ast_xml_node_get_next(event)) {
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struct ast_xml_node *event_instance;
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const char *name = ast_xml_get_attribute(event, "name");
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RAII_VAR(const char *, name, ast_xml_get_attribute(event, "name"),
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ast_xml_free_attr);
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struct ast_xml_doc_item *new_item;
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if (!name || strcmp(ast_xml_node_get_name(event), "managerEvent")) {
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@@ -2607,10 +2608,16 @@ static struct ast_xml_doc_item *xmldoc_build_final_response(struct ast_xml_node
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"managerEventInstance", NULL, NULL);
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if (!event_instance) {
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return NULL;
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} else {
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const char *name;
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struct ast_xml_doc_item *res;
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name = ast_xml_get_attribute(final_response_event, "name");
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res = xmldoc_build_documentation_item(event_instance, name, "managerEvent");
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ast_xml_free_attr(name);
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return res;
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}
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return xmldoc_build_documentation_item(event_instance,
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ast_xml_get_attribute(final_response_event, "name"), "managerEvent");
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}
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struct ast_xml_doc_item *ast_xmldoc_build_final_response(const char *type, const char *name, const char *module)
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@@ -377,14 +377,12 @@ static void *attempt_thread(void *data)
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res = ast_pbx_outgoing_app(o->tech, o->capabilities, o->dest, o->waittime * 1000,
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o->app, o->data, &reason, 2 /* wait to finish */, o->cid_num, o->cid_name,
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o->vars, o->account, NULL, NULL);
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o->vars = NULL;
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} else {
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ast_verb(3, "Attempting call on %s/%s for %s@%s:%d (Retry %d)\n", o->tech, o->dest, o->exten, o->context,o->priority, o->retries);
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res = ast_pbx_outgoing_exten(o->tech, o->capabilities, o->dest,
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o->waittime * 1000, o->context, o->exten, o->priority, &reason,
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2 /* wait to finish */, o->cid_num, o->cid_name, o->vars, o->account, NULL,
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ast_test_flag(&o->options, SPOOL_FLAG_EARLY_MEDIA), NULL);
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o->vars = NULL;
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}
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if (res) {
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ast_log(LOG_NOTICE, "Call failed to go through, reason (%d) %s\n", reason, ast_channel_reason2str(reason));
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@@ -242,16 +242,16 @@ static struct ast_frame *refer_progress_framehook(struct ast_channel *chan, stru
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/* If a notification is due to be sent push it to the thread pool */
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if (notification) {
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if (ast_sip_push_task(progress->serializer, refer_progress_notify, notification)) {
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ao2_cleanup(notification);
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}
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/* If the subscription is being terminated we don't need the frame hook any longer */
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if (notification->state == PJSIP_EVSUB_STATE_TERMINATED) {
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ast_debug(3, "Detaching REFER progress monitoring hook from '%s' as subscription is being terminated\n",
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ast_channel_name(chan));
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ast_framehook_detach(chan, progress->framehook);
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}
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if (ast_sip_push_task(progress->serializer, refer_progress_notify, notification)) {
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ao2_cleanup(notification);
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}
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}
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return f;
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@@ -420,6 +420,7 @@ static void refer_attended_destroy(void *obj)
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ao2_cleanup(attended->transferer);
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ast_channel_unref(attended->transferer_chan);
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ao2_cleanup(attended->transferer_second);
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ao2_cleanup(attended->progress);
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}
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/*! \brief Allocator for attended transfer task */
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