| CVE |
Vendors |
Products |
Updated |
CVSS v3.1 |
| chan_iax2.c in the IAX2 channel driver in Asterisk Open Source 1.4.x before 1.4.41.1, 1.6.2.x before 1.6.2.18.1, and 1.8.x before 1.8.4.3, and Asterisk Business Edition C.3 before C.3.7.3, accesses a memory address contained in an option control frame, which allows remote attackers to cause a denial of service (daemon crash) or possibly have unspecified other impact via a crafted frame. |
| tcptls.c in the TCP/TLS server in Asterisk Open Source 1.6.1.x before 1.6.1.23, 1.6.2.x before 1.6.2.17.1, and 1.8.x before 1.8.3.1 allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) by establishing many short TCP sessions to services that use a certain TLS API. |
| 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. |
| Asterisk Open Source 1.8.x before 1.8.19.1, 10.x before 10.11.1, and 11.x before 11.1.2; Certified Asterisk 1.8.11 before 1.8.11-cert10; and Asterisk Digiumphones 10.x-digiumphones before 10.11.1-digiumphones, when anonymous calls are enabled, allow remote attackers to cause a denial of service (resource consumption) by making anonymous calls from multiple sources and consequently adding many entries to the device state cache. |
| Asterisk Open Source 1.4.x before 1.4.40.1, 1.6.1.x before 1.6.1.25, 1.6.2.x before 1.6.2.17.3, and 1.8.x before 1.8.3.3 and Asterisk Business Edition C.x.x before C.3.6.4 do not restrict the number of unauthenticated sessions to certain interfaces, which allows remote attackers to cause a denial of service (file descriptor exhaustion and disk space exhaustion) via a series of TCP connections. |
| chan_sip.c in the SIP channel driver in Asterisk Open Source 1.6.x before 1.6.2.18.1 and 1.8.x before 1.8.4.3 does not properly handle '\0' characters in SIP packets, which allows remote attackers to cause a denial of service (memory corruption) or possibly have unspecified other impact via a crafted packet. |
| The SIP over UDP implementation in Asterisk Open Source 1.4.x before 1.4.43, 1.6.x before 1.6.2.21, and 1.8.x before 1.8.7.2 uses different port numbers for responses to invalid requests depending on whether a SIP username exists, which allows remote attackers to enumerate usernames via a series of requests. |
| Asterisk is an open source private branch exchange and telephony toolkit. In Asterisk prior to versions 18.20.1, 20.5.1, and 21.0.1; as well as certified-asterisk prior to 18.9-cert6; Asterisk is susceptible to a DoS due to a race condition in the hello handshake phase of the DTLS protocol when handling DTLS-SRTP for media setup. This attack can be done continuously, thus denying new DTLS-SRTP encrypted calls during the attack. Abuse of this vulnerability may lead to a massive Denial of Service on vulnerable Asterisk servers for calls that rely on DTLS-SRTP. Commit d7d7764cb07c8a1872804321302ef93bf62cba05 contains a fix, which is part of versions 18.20.1, 20.5.1, 21.0.1, amd 18.9-cert6. |
| Asterisk is an open source private branch exchange and telephony toolkit. In Asterisk prior to versions 18.20.1, 20.5.1, and 21.0.1, as well as certified-asterisk prior to 18.9-cert6, it is possible to read any arbitrary file even when the `live_dangerously` is not enabled. This allows arbitrary files to be read. Asterisk versions 18.20.1, 20.5.1, and 21.0.1, as well as certified-asterisk prior to 18.9-cert6, contain a fix for this issue. |
| Asterisk is an open source private branch exchange and telephony toolkit. In Asterisk versions 18.20.0 and prior, 20.5.0 and prior, and 21.0.0; as well as ceritifed-asterisk 18.9-cert5 and prior, the 'update' functionality of the PJSIP_HEADER dialplan function can exceed the available buffer space for storing the new value of a header. By doing so this can overwrite memory or cause a crash. This is not externally exploitable, unless dialplan is explicitly written to update a header based on data from an outside source. If the 'update' functionality is not used the vulnerability does not occur. A patch is available at commit a1ca0268254374b515fa5992f01340f7717113fa. |
| An issue was discovered in Asterisk through 19.x and Certified Asterisk through 16.8-cert13. The func_odbc module provides possibly inadequate escaping functionality for backslash characters in SQL queries, resulting in user-provided data creating a broken SQL query or possibly a SQL injection. This is fixed in 16.25.2, 18.11.2, and 19.3.2, and 16.8-cert14. |
| An SSRF issue was discovered in Asterisk through 19.x. When using STIR/SHAKEN, it's possible to send arbitrary requests (such as GET) to interfaces such as localhost by using the Identity header. This is fixed in 16.25.2, 18.11.2, and 19.3.2. |
| An issue was discovered in Asterisk through 19.x. When using STIR/SHAKEN, it is possible to download files that are not certificates. These files could be much larger than what one would expect to download, leading to Resource Exhaustion. This is fixed in 16.25.2, 18.11.2, and 19.3.2. |
| res_pjsip_t38 in Sangoma Asterisk 16.x before 16.16.2, 17.x before 17.9.3, and 18.x before 18.2.2, and Certified Asterisk before 16.8-cert7, allows an attacker to trigger a crash by sending an m=image line and zero port in a response to a T.38 re-invite initiated by Asterisk. This is a re-occurrence of the CVE-2019-15297 symptoms but not for exactly the same reason. The crash occurs because there is an append operation relative to the active topology, but this should instead be a replace operation. |
| 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. |
