Asterisk » Open Source : Security Vulnerabilities, CVEs,

A res_pjsip_session crash was discovered in Asterisk Open Source 13.x before 13.37.1, 16.x before 16.14.1, 17.x before 17.8.1, and 18.x before 18.0.1. and Certified Asterisk before 16.8-cert5. Upon receiving a new SIP Invite, Asterisk did not return the created dialog locked or referenced. This caused a gap between the creation of the dialog object, and its next use by the thread that created it. Depending on some off-nominal circumstances and timing, it was possible for another thread to free said dialog in this gap. Asterisk could then crash when the dialog object, or any of its dependent objects, were dereferenced or accessed next by the initial-creation thread. Note, however, that this crash can only occur when using a connection-oriented protocol (e.g., TCP or TLS, but not UDP) for SIP transport. Also, the remote client must be authenticated, or Asterisk must be configured for anonymous calling.

An issue was discovered in Asterisk Open Source 13.x before 13.37.1, 16.x before 16.14.1, 17.x before 17.8.1, and 18.x before 18.0.1 and Certified Asterisk before 16.8-cert5. If Asterisk is challenged on an outbound INVITE and the nonce is changed in each response, Asterisk will continually send INVITEs in a loop. This causes Asterisk to consume more and more memory since the transaction will never terminate (even if the call is hung up), ultimately leading to a restart or shutdown of Asterisk. Outbound authentication must be configured on the endpoint for this to occur.

An issue was discovered in Asterisk Open Source 15.x before 15.4.1. When connected to Asterisk via TCP/TLS, if the client abruptly disconnects, or sends a specially crafted message, then Asterisk gets caught in an infinite loop while trying to read the data stream. This renders the system unusable.

A memory exhaustion vulnerability exists in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1 and Certified Asterisk 13.13 before 13.13-cert4, which can be triggered by sending specially crafted SCCP packets causing an infinite loop and leading to memory exhaustion (by message logging in that loop).

main/http.c in the HTTP server in Asterisk Open Source 1.8.x before 1.8.20.2, 10.x before 10.12.2, and 11.x before 11.2.2; Certified Asterisk 1.8.15 before 1.8.15-cert2; and Asterisk Digiumphones 10.x-digiumphones before 10.12.2-digiumphones does not properly restrict Content-Length values, which allows remote attackers to conduct stack-consumption attacks and cause a denial of service (daemon crash) via a crafted HTTP POST request. NOTE: this vulnerability exists because of an incorrect fix for CVE-2012-5976.

Stack-based buffer overflow in res/res_format_attr_h264.c in Asterisk Open Source 11.x before 11.2.2 allows remote attackers to execute arbitrary code via a long sprop-parameter-sets H.264 media attribute in a SIP Session Description Protocol (SDP) header.

The SIP channel driver in Asterisk Open Source 1.8.x before 1.8.20.2, 10.x before 10.12.2, and 11.x before 11.2.2; Certified Asterisk 1.8.15 before 1.8.15-cert2; Asterisk Business Edition (BE) C.3.x before C.3.8.1; and Asterisk Digiumphones 10.x-digiumphones before 10.12.2-digiumphones exhibits different behavior for invalid INVITE, SUBSCRIBE, and REGISTER transactions depending on whether the user account exists, which allows remote attackers to enumerate account names by (1) reading HTTP status codes, (2) reading additional text in a 403 (aka Forbidden) response, or (3) observing whether certain retransmissions occur.

chan_skinny.c in the Skinny (aka SCCP) channel driver in Certified Asterisk 1.8.11-cert before 1.8.11-cert2 and Asterisk Open Source 1.8.x before 1.8.12.1 and 10.x before 10.4.1 allows remote authenticated users to cause a denial of service (NULL pointer dereference and daemon crash) by closing a connection in off-hook mode.

chan_sip.c in the SIP channel driver in Asterisk Open Source 1.8.x before 1.8.11.1 and 10.x before 10.3.1 and Asterisk Business Edition C.3.x before C.3.7.4, when the trustrpid option is enabled, allows remote authenticated users to cause a denial of service (daemon crash) by sending a SIP UPDATE message that triggers a connected-line update attempt without an associated channel.

Heap-based buffer overflow in chan_skinny.c in the Skinny channel driver in Asterisk Open Source 1.6.2.x before 1.6.2.24, 1.8.x before 1.8.11.1, and 10.x before 10.3.1 allows remote authenticated users to cause a denial of service or possibly have unspecified other impact via a series of KEYPAD_BUTTON_MESSAGE events.

main/manager.c in the Manager Interface in Asterisk Open Source 1.6.2.x before 1.6.2.24, 1.8.x before 1.8.11.1, and 10.x before 10.3.1 and Asterisk Business Edition C.3.x before C.3.7.4 does not properly enforce System class authorization requirements, which allows remote authenticated users to execute arbitrary commands via (1) the originate action in the MixMonitor application, (2) the SHELL and EVAL functions in the GetVar manager action, or (3) the SHELL and EVAL functions in the Status manager action.

Incomplete blacklist vulnerability in main/manager.c in Asterisk Open Source 1.8.x before 1.8.15.1 and 10.x before 10.7.1, Certified Asterisk 1.8.11 before 1.8.11-cert6, Asterisk Digiumphones 10.x.x-digiumphones before 10.7.1-digiumphones, and Asterisk Business Edition C.3.x before C.3.7.6 allows remote authenticated users to execute arbitrary commands by leveraging originate privileges and providing an ExternalIVR value in an AMI Originate action.

chan_sip.c in Asterisk Open Source 1.8.x before 1.8.8.2 and 10.x before 10.0.1, when the res_srtp module is used and media support is improperly configured, allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a crafted SDP message with a crypto attribute and a (1) video or (2) text media type, as demonstrated by CSipSimple.

chan_sip.c in the SIP channel driver in Asterisk Open Source 1.8.x before 1.8.7.1 and 10.x before 10.0.0-rc1 does not properly initialize variables during request parsing, which allows remote authenticated users to cause a denial of service (daemon crash) via a malformed request.

asterisk allows calls on prohibited networks

The IAX2 protocol implementation in Asterisk Open Source 1.2.x before 1.2.35, 1.4.x before 1.4.26.2, 1.6.0.x before 1.6.0.15, and 1.6.1.x before 1.6.1.6; Business Edition B.x.x before B.2.5.10, C.2.x before C.2.4.3, and C.3.x before C.3.1.1; and s800i 1.3.x before 1.3.0.3 allows remote attackers to cause a denial of service (call-number exhaustion) by initiating many IAX2 message exchanges, a related issue to CVE-2008-3263.

IAX2 in Asterisk Open Source 1.2.x before 1.2.31, 1.4.x before 1.4.23-rc4, and 1.6.x before 1.6.0.3-rc2; Business Edition A.x.x, B.x.x before B.2.5.7, C.1.x.x before C.1.10.4, and C.2.x.x before C.2.1.2.1; and s800i 1.2.x before 1.3.0 responds differently to a failed login attempt depending on whether the user account exists, which allows remote attackers to enumerate valid usernames.

Asterisk Open Source 1.2.26 through 1.2.30.3 and Business Edition B.2.3.5 through B.2.5.5, when realtime IAX2 users are enabled, allows remote attackers to cause a denial of service (crash) via authentication attempts involving (1) an unknown user or (2) a user using hostname matching.

The FWDOWNL firmware-download implementation in Asterisk Open Source 1.0.x, 1.2.x before 1.2.30, and 1.4.x before 1.4.21.2; Business Edition A.x.x, B.x.x before B.2.5.4, and C.x.x before C.1.10.3; AsteriskNOW; Appliance Developer Kit 0.x.x; and s800i 1.0.x before 1.2.0.1 allows remote attackers to cause a denial of service (traffic amplification) via an IAX2 FWDOWNL request.

Asterisk Open Source 1.0.x and 1.2.x before 1.2.29 and Business Edition A.x.x and B.x.x before B.2.5.3, when pedantic parsing (aka pedanticsipchecking) is enabled, allows remote attackers to cause a denial of service (daemon crash) via a SIP INVITE message that lacks a From header, related to invocations of the ast_uri_decode function, and improper handling of (1) an empty const string and (2) a NULL pointer.

The IAX2 channel driver (chan_iax2) in Asterisk 1.2 before revision 72630 and 1.4 before revision 65679, when configured to allow unauthenticated calls, sends "early audio" to an unverified source IP address of a NEW message, which allows remote attackers to cause a denial of service (traffic amplification) via a spoofed NEW message.

The IAX2 channel driver (chan_iax2) in Asterisk Open Source 1.0.x, 1.2.x before 1.2.28, and 1.4.x before 1.4.19.1; Business Edition A.x.x, B.x.x before B.2.5.2, and C.x.x before C.1.8.1; AsteriskNOW before 1.0.3; Appliance Developer Kit 0.x.x; and s800i before 1.1.0.3, when configured to allow unauthenticated calls, does not verify that an ACK response contains a call number matching the server's reply to a NEW message, which allows remote attackers to cause a denial of service (traffic amplification) via a spoofed ACK response that does not complete a 3-way handshake. NOTE: this issue exists because of an incomplete fix for CVE-2008-1923.

Format string vulnerability in Asterisk Open Source 1.6.x before 1.6.0-beta6 might allow remote attackers to execute arbitrary code via logging messages that are not properly handled by (1) the ast_verbose logging API call, or (2) the astman_append function.

Unspecified vulnerability in Asterisk Open Source 1.2.x before 1.2.27, 1.4.x before 1.4.18.1 and 1.4.19-rc3; Business Edition A.x.x, B.x.x before B.2.5.1, and C.x.x before C.1.6.2; AsteriskNOW 1.0.x before 1.0.2; Appliance Developer Kit before 1.4 revision 109393; and s800i 1.0.x before 1.1.0.2; allows remote attackers to access the SIP channel driver via a crafted From header.

Multiple buffer overflows in Asterisk Open Source 1.4.x before 1.4.18.1 and 1.4.19-rc3, Open Source 1.6.x before 1.6.0-beta6, Business Edition C.x.x before C.1.6.1, AsteriskNOW 1.0.x before 1.0.2, Appliance Developer Kit before 1.4 revision 109386, and s800i 1.1.x before 1.1.0.2 allow remote attackers to (1) write a zero to an arbitrary memory location via a large RTP payload number, related to the ast_rtp_unset_m_type function in main/rtp.c; or (2) write certain integers to an arbitrary memory location via a large number of RTP payloads, related to the process_sdp function in channels/chan_sip.c.