[asterisk-commits] res pjsip refer: Fix crash from a REFER and BYE collision. (asterisk[certified/13.1])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jun 2 17:23:22 CDT 2015
Joshua Colp has submitted this change and it was merged.
Change subject: res_pjsip_refer: Fix crash from a REFER and BYE collision.
......................................................................
res_pjsip_refer: Fix crash from a REFER and BYE collision.
Analyzing a one-off crash on a busy system showed that processing a REFER
request had a NULL session channel pointer. The only way I can think of
that could cause this is if an outgoing BYE transaction overlapped the
incoming REFER transaction in a collision. Asterisk sends a BYE while the
phone sends a REFER to complete an attended transfer.
* Made check the session channel pointer before processing an incoming
REFER request in res_pjsip_refer.
* Fixed similar crash potential for res_pjsip supplement incoming request
processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE,
res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER
messages.
* Made res_pjsip_messaging respond to a message body too large with a 413
instead of ignoring it.
ASTERISK-24700 #close
Reported by: Zane Conkle
Review: https://reviewboard.asterisk.org/r/4417/
........
Merged revisions 431898 from http://svn.asterisk.org/svn/asterisk/branches/13
Change-Id: I57878adc0846dd942a699ad36dcec9cba5e57994
---
M res/res_pjsip_caller_id.c
M res/res_pjsip_messaging.c
M res/res_pjsip_refer.c
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_send_to_voicemail.c
5 files changed, 31 insertions(+), 15 deletions(-)
Approvals:
Joshua Colp: Looks good to me, approved; Verified
diff --git a/res/res_pjsip_caller_id.c b/res/res_pjsip_caller_id.c
index c3757e0..dc595c4 100644
--- a/res/res_pjsip_caller_id.c
+++ b/res/res_pjsip_caller_id.c
@@ -361,7 +361,7 @@
if (!session->endpoint->id.self.number.valid) {
set_id_from_from(rdata, &session->id);
}
- } else {
+ } else if (session->channel) {
/* Reinvite. Check for changes to the ID and queue a connected line
* update if necessary
*/
diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
index c125864..dab70ca 100644
--- a/res/res_pjsip_messaging.c
+++ b/res/res_pjsip_messaging.c
@@ -692,9 +692,13 @@
char buf[MAX_BODY_SIZE];
enum pjsip_status_code code;
struct ast_frame f;
-
pjsip_dialog *dlg = session->inv_session->dlg;
pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
+
+ if (!session->channel) {
+ send_response(rdata, PJSIP_SC_NOT_FOUND, dlg, tsx);
+ return 0;
+ }
if ((code = check_content_type(rdata)) != PJSIP_SC_OK) {
send_response(rdata, code, dlg, tsx);
@@ -703,6 +707,7 @@
if (print_body(rdata, buf, sizeof(buf)-1) < 1) {
/* invalid body size */
+ send_response(rdata, PJSIP_SC_REQUEST_ENTITY_TOO_LARGE, dlg, tsx);
return 0;
}
diff --git a/res/res_pjsip_refer.c b/res/res_pjsip_refer.c
index 88f135c..771ab11 100644
--- a/res/res_pjsip_refer.c
+++ b/res/res_pjsip_refer.c
@@ -418,7 +418,7 @@
struct refer_attended *attended = obj;
ao2_cleanup(attended->transferer);
- ast_channel_unref(attended->transferer_chan);
+ ast_channel_cleanup(attended->transferer_chan);
ao2_cleanup(attended->transferer_second);
ao2_cleanup(attended->progress);
}
@@ -667,7 +667,7 @@
return 200;
} else {
- const char *context = (session->channel ? pbx_builtin_getvar_helper(session->channel, "TRANSFER_CONTEXT") : "");
+ const char *context = pbx_builtin_getvar_helper(session->channel, "TRANSFER_CONTEXT");
struct refer_blind refer = { 0, };
if (ast_strlen_zero(context)) {
@@ -710,10 +710,6 @@
const char *context;
char exten[AST_MAX_EXTENSION];
struct refer_blind refer = { 0, };
-
- if (!session->channel) {
- return 404;
- }
/* If no explicit transfer context has been provided use their configured context */
context = pbx_builtin_getvar_helper(session->channel, "TRANSFER_CONTEXT");
@@ -886,6 +882,14 @@
static const pj_str_t str_refer_to = { "Refer-To", 8 };
static const pj_str_t str_replaces = { "Replaces", 8 };
+ if (!session->channel) {
+ /* No channel to refer. Likely because the call was just hung up. */
+ pjsip_dlg_respond(session->inv_session->dlg, rdata, 404, NULL, NULL, NULL);
+ ast_debug(3, "Received a REFER on a session with no channel from endpoint '%s'.\n",
+ ast_sorcery_object_get_id(session->endpoint));
+ return 0;
+ }
+
if (!session->endpoint->allowtransfer) {
pjsip_dlg_respond(session->inv_session->dlg, rdata, 603, NULL, NULL, NULL);
ast_log(LOG_WARNING, "Endpoint %s transfer attempt blocked due to configuration\n",
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 1994db7..a85d1e8 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -1274,15 +1274,18 @@
static int video_info_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
- struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
+ struct pjsip_transaction *tsx;
pjsip_tx_data *tdata;
- if (!ast_sip_is_content_type(&rdata->msg_info.msg->body->content_type,
- "application",
- "media_control+xml")) {
+ if (!session->channel
+ || !ast_sip_is_content_type(&rdata->msg_info.msg->body->content_type,
+ "application",
+ "media_control+xml")) {
return 0;
}
+ tsx = pjsip_rdata_get_tsx(rdata);
+
ast_queue_control(session->channel, AST_CONTROL_VIDUPDATE);
if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, 200, NULL, &tdata) == PJ_SUCCESS) {
diff --git a/res/res_pjsip_send_to_voicemail.c b/res/res_pjsip_send_to_voicemail.c
index 97f55d3..3a57aea 100644
--- a/res/res_pjsip_send_to_voicemail.c
+++ b/res/res_pjsip_send_to_voicemail.c
@@ -119,13 +119,17 @@
static int handle_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
-
struct ast_datastore *sip_session_datastore;
struct ast_channel *other_party;
+ int has_feature;
+ int has_reason;
- int has_feature = has_call_feature(rdata);
- int has_reason = has_diversion_reason(rdata);
+ if (!session->channel) {
+ return 0;
+ }
+ has_feature = has_call_feature(rdata);
+ has_reason = has_diversion_reason(rdata);
if (!has_feature && !has_reason) {
/* If we don't have a call feature or diversion reason or if
it's not a feature this module is related to then there
--
To view, visit https://gerrit.asterisk.org/566
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Gerrit-MessageType: merged
Gerrit-Change-Id: I57878adc0846dd942a699ad36dcec9cba5e57994
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: certified/13.1
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
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