| CVE |
Vendors |
Products |
Updated |
CVSS v3.1 |
| manager.c in Asterisk Open Source 1.6.1.x before 1.6.1.24, 1.6.2.x before 1.6.2.17.2, and 1.8.x before 1.8.3.2 allows remote attackers to cause a denial of service (CPU and memory consumption) via a series of manager sessions involving invalid data. |
| Multiple stack-based and heap-based buffer overflows in the (1) decode_open_type and (2) udptl_rx_packet functions in main/udptl.c in Asterisk Open Source 1.4.x before 1.4.39.2, 1.6.1.x before 1.6.1.22, 1.6.2.x before 1.6.2.16.2, and 1.8 before 1.8.2.4; Business Edition C.x.x before C.3.6.3; AsteriskNOW 1.5; and s800i (Asterisk Appliance), when T.38 support is enabled, allow remote attackers to cause a denial of service (crash) and possibly execute arbitrary code via a crafted UDPTL packet. |
| Stack-based buffer overflow in the ast_uri_encode function in main/utils.c in Asterisk Open Source before 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.1, 1.8.1.2, 1.8.2.; and Business Edition before C.3.6.2; when running in pedantic mode allows remote authenticated users to execute arbitrary code via crafted caller ID data in vectors involving the (1) SIP channel driver, (2) URIENCODE dialplan function, or (3) AGI dialplan function. |
| main/acl.c in Asterisk Open Source 1.6.0.x before 1.6.0.25, 1.6.1.x before 1.6.1.17, and 1.6.2.x before 1.6.2.5 does not properly enforce remote host access controls when CIDR notation "/0" is used in permit= and deny= configuration rules, which causes an improper arithmetic shift and might allow remote attackers to bypass ACL rules and access services from unauthorized hosts. |
| An issue was discovered in Sangoma Asterisk 13.x before 13.38.3, 16.x before 16.19.1, 17.x before 17.9.4, and 18.x before 18.5.1, and Certified Asterisk before 16.8-cert10. If the IAX2 channel driver receives a packet that contains an unsupported media format, a crash can occur. |
| An issue was discovered in PJSIP in Asterisk before 16.19.1 and before 18.5.1. To exploit, a re-INVITE without SDP must be received after Asterisk has sent a BYE request. |
| An issue was discovered in res_pjsip_session.c in Digium Asterisk through 13.38.1; 14.x, 15.x, and 16.x through 16.16.0; 17.x through 17.9.1; and 18.x through 18.2.0, and Certified Asterisk through 16.8-cert5. An SDP negotiation vulnerability in PJSIP allows a remote server to potentially crash Asterisk by sending specific SIP responses that cause an SDP negotiation failure. |
| An issue was discovered in Sangoma Asterisk 16.x before 16.16.1, 17.x before 17.9.2, and 18.x before 18.2.1 and Certified Asterisk before 16.8-cert6. When re-negotiating for T.38, if the initial remote response was delayed just enough, Asterisk would send both audio and T.38 in the SDP. If this happened, and the remote responded with a declined T.38 stream, then Asterisk would crash. |
| A stack-based buffer overflow in res_rtp_asterisk.c in Sangoma Asterisk before 16.16.1, 17.x before 17.9.2, and 18.x before 18.2.1 and Certified Asterisk before 16.8-cert6 allows an authenticated WebRTC client to cause an Asterisk crash by sending multiple hold/unhold requests in quick succession. This is caused by a signedness comparison mismatch. |
| Incorrect access controls in res_srtp.c in Sangoma Asterisk 13.38.1, 16.16.0, 17.9.1, and 18.2.0 and Certified Asterisk 16.8-cert5 allow a remote unauthenticated attacker to prematurely terminate secure calls by replaying SRTP packets. |
| A buffer overflow in res_pjsip_diversion.c in Sangoma Asterisk versions 13.38.1, 16.15.1, 17.9.1, and 18.1.1 allows remote attacker to crash Asterisk by deliberately misusing SIP 181 responses. |
| An issue was discovered in res_pjsip_diversion.c in Sangoma Asterisk before 13.38.0, 14.x through 16.x before 16.15.0, 17.x before 17.9.0, and 18.x before 18.1.0. A crash can occur when a SIP message is received with a History-Info header that contains a tel-uri, or when a SIP 181 response is received that contains a tel-uri in the Diversion header. |
| 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 Integer Signedness issue (for a return code) in the res_pjsip_sdp_rtp module in Digium Asterisk versions 15.7.1 and earlier and 16.1.1 and earlier allows remote authenticated users to crash Asterisk via a specially crafted SDP protocol violation. |
| An issue was discovered in res_pjsip_t38.c in Sangoma Asterisk through 13.x and Certified Asterisk through 13.21-x. If it receives a re-invite initiating T.38 faxing and has a port of 0 and no c line in the SDP, a NULL pointer dereference and crash will occur. This is different from CVE-2019-18940. |
| An issue was discovered in channels/chan_sip.c in Sangoma Asterisk 13.x before 13.29.2, 16.x before 16.6.2, and 17.x before 17.0.1, and Certified Asterisk 13.21 before cert5. A SIP request can be sent to Asterisk that can change a SIP peer's IP address. A REGISTER does not need to occur, and calls can be hijacked as a result. The only thing that needs to be known is the peer's name; authentication details such as passwords do not need to be known. This vulnerability is only exploitable when the nat option is set to the default, or auto_force_rport. |
| An issue was discovered in manager.c in Sangoma Asterisk through 13.x, 16.x, 17.x and Certified Asterisk 13.21 through 13.21-cert4. A remote authenticated Asterisk Manager Interface (AMI) user without system authorization could use a specially crafted Originate AMI request to execute arbitrary system commands. |
| main/translate.c in Sangoma Asterisk 13.28.0 and 16.5.0 allows a remote attacker to send a specific RTP packet during a call and cause a crash in a specific scenario. |
| res_pjsip_t38 in Sangoma Asterisk 15.x before 15.7.4 and 16.x before 16.5.1 allows an attacker to trigger a crash by sending a declined stream in a response to a T.38 re-invite initiated by Asterisk. The crash occurs because of a NULL session media object dereference. |
| An issue was discovered in Asterisk Open Source through 13.27.0, 14.x and 15.x through 15.7.2, and 16.x through 16.4.0, and Certified Asterisk through 13.21-cert3. A pointer dereference in chan_sip while handling SDP negotiation allows an attacker to crash Asterisk when handling an SDP answer to an outgoing T.38 re-invite. To exploit this vulnerability an attacker must cause the chan_sip module to send a T.38 re-invite request to them. Upon receipt, the attacker must send an SDP answer containing both a T.38 UDPTL stream and another media stream containing only a codec (which is not permitted according to the chan_sip configuration). |
