| CVE |
Vendors |
Products |
Updated |
CVSS v3.1 |
| 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. |
| The AsteriskGUI HTTP server in Asterisk Open Source 1.4.x before 1.4.19-rc3 and 1.6.x before 1.6.0-beta6, Business Edition C.x.x before C.1.6, AsteriskNOW before 1.0.2, Appliance Developer Kit before revision 104704, and s800i 1.0.x before 1.1.0.2 generates insufficiently random manager ID values, which makes it easier for remote attackers to hijack a manager session via a series of ID guesses. |
| 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 1.0.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. |
| Multiple stack-based buffer overflows in the process_sdp function in chan_sip.c of the SIP channel T.38 SDP parser in Asterisk before 1.4.3 allow remote attackers to execute arbitrary code via a long (1) T38FaxRateManagement or (2) T38FaxUdpEC SDP parameter in an SIP message, as demonstrated using SIP INVITE. |
| The Manager Interface in Asterisk before 1.2.18 and 1.4.x before 1.4.3 allows remote attackers to cause a denial of service (crash) by using MD5 authentication to authenticate a user that does not have a password defined in manager.conf, resulting in a NULL pointer dereference. |
| 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. |
| PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In affected versions if the incoming STUN message contains an ERROR-CODE attribute, the header length is not checked before performing a subtraction operation, potentially resulting in an integer underflow scenario. This issue affects all users that use STUN. A malicious actor located within the victim’s network may forge and send a specially crafted UDP (STUN) message that could remotely execute arbitrary code on the victim’s machine. Users are advised to upgrade as soon as possible. There are no known workarounds. |
| 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. |
| Asterisk is an open source private branch exchange (PBX) and telephony toolkit. Prior to asterisk versions 18.24.2, 20.9.2, and 21.4.2 and certified-asterisk versions 18.9-cert11 and 20.7-cert2, an AMI user with `write=originate` may change all configuration files in the `/etc/asterisk/` directory. This occurs because they are able to curl remote files and write them to disk, but are also able to append to existing files using the `FILE` function inside the `SET` application. This issue may result in privilege escalation, remote code execution and/or blind server-side request forgery with arbitrary protocol. Asterisk versions 18.24.2, 20.9.2, and 21.4.2 and certified-asterisk versions 18.9-cert11 and 20.7-cert2 contain a fix for this issue. |
