diff options
author | Richard Mudgett <rmudgett@digium.com> | 2015-02-17 15:31:46 +0000 |
---|---|---|
committer | Richard Mudgett <rmudgett@digium.com> | 2015-02-17 15:31:46 +0000 |
commit | 6d3fcfc3c2f49b3909b7ae0ebb74d99e2fedbb65 (patch) | |
tree | 8238044471bb69f8a8ae40eeeb41a3af3771b9da /res | |
parent | 562b7bf6f09d9ea5ac8e20575d87f4e892609c20 (diff) |
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res')
-rw-r--r-- | res/res_pjsip_caller_id.c | 2 | ||||
-rw-r--r-- | res/res_pjsip_messaging.c | 7 | ||||
-rw-r--r-- | res/res_pjsip_refer.c | 16 | ||||
-rw-r--r-- | res/res_pjsip_sdp_rtp.c | 11 | ||||
-rw-r--r-- | res/res_pjsip_send_to_voicemail.c | 10 |
5 files changed, 31 insertions, 15 deletions
diff --git a/res/res_pjsip_caller_id.c b/res/res_pjsip_caller_id.c index c3757e06f..dc595c4d8 100644 --- a/res/res_pjsip_caller_id.c +++ b/res/res_pjsip_caller_id.c @@ -361,7 +361,7 @@ static int caller_id_incoming_request(struct ast_sip_session *session, pjsip_rx_ 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 04332996a..813090816 100644 --- a/res/res_pjsip_messaging.c +++ b/res/res_pjsip_messaging.c @@ -681,10 +681,14 @@ static int incoming_in_dialog_request(struct ast_sip_session *session, struct pj 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); return 0; @@ -692,6 +696,7 @@ static int incoming_in_dialog_request(struct ast_sip_session *session, struct pj 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 cc0616e9d..b0755b1ea 100644 --- a/res/res_pjsip_refer.c +++ b/res/res_pjsip_refer.c @@ -418,7 +418,7 @@ static void refer_attended_destroy(void *obj) 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); } @@ -674,7 +674,7 @@ static int refer_incoming_attended_request(struct ast_sip_session *session, pjsi 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)) { @@ -718,10 +718,6 @@ static int refer_incoming_blind_request(struct ast_sip_session *session, pjsip_r 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"); if (ast_strlen_zero(context)) { @@ -893,6 +889,14 @@ static int refer_incoming_refer_request(struct ast_sip_session *session, struct 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 6471341b3..63a78caa8 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -1268,15 +1268,18 @@ static struct ast_sip_session_sdp_handler video_sdp_handler = { 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 97f55d300..3a57aea7a 100644 --- a/res/res_pjsip_send_to_voicemail.c +++ b/res/res_pjsip_send_to_voicemail.c @@ -119,13 +119,17 @@ static int has_call_feature(pjsip_rx_data *rdata) 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 |