| CVE |
Vendors |
Products |
Updated |
CVSS v3.1 |
| The handle_response function in chan_sip.c in Asterisk before 1.2.17 and 1.4.x before 1.4.2 allows remote attackers to cause a denial of service (crash) via a SIP Response code 0 in a SIP packet. |
| The Asterisk Extension Language (AEL) in pbx/pbx_ael.c in Asterisk does not properly generate extensions, which allows remote attackers to execute arbitrary extensions and have an unknown impact by specifying an invalid extension in a certain form. |
| 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. |
| The SIP channel driver (chan_sip) in Asterisk before 1.2.18 and 1.4.x before 1.4.3 does not properly parse SIP UDP packets that do not contain a valid response code, which allows remote attackers to cause a denial of service (crash). |
| Stack-based buffer overflow in the IAX2 channel driver (chan_iax2) in Asterisk before 1.2.22 and 1.4.x before 1.4.8, Business Edition before B.2.2.1, AsteriskNOW before beta7, Appliance Developer Kit before 0.5.0, and s800i before 1.0.2 allows remote attackers to execute arbitrary code by sending a long (1) voice or (2) video RTP frame. |
| The IAX2 channel driver (chan_iax2) in Asterisk before 1.2.22 and 1.4.x before 1.4.8, Business Edition before B.2.2.1, AsteriskNOW before beta7, Appliance Developer Kit before 0.5.0, and s800i before 1.0.2 allows remote attackers to cause a denial of service (crash) via a crafted (1) LAGRQ or (2) LAGRP frame that contains information elements of IAX frames, which results in a NULL pointer dereference when Asterisk does not properly set an associated variable. |
| The Skinny channel driver (chan_skinny) in Asterisk before 1.2.22 and 1.4.x before 1.4.8, Business Edition before B.2.2.1, AsteriskNOW before beta7, Appliance Developer Kit before 0.5.0, and s800i before 1.0.2 allows remote attackers to cause a denial of service (crash) via a certain data length value in a crafted packet, which results in an "overly large memcpy." |
| The STUN implementation in Asterisk 1.4.x before 1.4.8, AsteriskNOW before beta7, Appliance Developer Kit before 0.5.0, and s800i before 1.0.2 allows remote attackers to cause a denial of service (crash) via a crafted STUN length attribute in a STUN packet sent on an RTP port. |
| A local privilege escalation vulnerability exists in the safe_asterisk script included with the Asterisk toolkit package. When Asterisk is started via this script (common in SysV init or FreePBX environments), it sources all .sh files located in /etc/asterisk/startup.d/ as root, without validating ownership or permissions.
Non-root users with legitimate write access to /etc/asterisk can exploit this behaviour by placing malicious scripts in the startup.d directory, which will then execute with root privileges upon service restart. |
| 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 versions up to and including 2.11.1 when in a dialog set (or forking) scenario, a hash key shared by multiple UAC dialogs can potentially be prematurely freed when one of the dialogs is destroyed . The issue may cause a dialog set to be registered in the hash table multiple times (with different hash keys) leading to undefined behavior such as dialog list collision which eventually leading to endless loop. A patch is available in commit db3235953baa56d2fb0e276ca510fefca751643f which will be included in the next release. There are no known workarounds for this issue. |
| 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 versions 2.11.1 and prior, parsing an incoming SIP message that contains a malformed multipart can potentially cause out-of-bound read access. This issue affects all PJSIP users that accept SIP multipart. The patch is available as commit in the `master` branch. There are no known workarounds. |
| 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. |
| Asterisk is an open-source private branch exchange (PBX). Prior to versions 18.24.3, 20.9.3, and 21.4.3 of Asterisk and versions 18.9-cert12 and 20.7-cert2 of certified-asterisk, if Asterisk attempts to send a SIP request to a URI whose host portion starts with `.1` or `[.1]`, and res_resolver_unbound is loaded, Asterisk will crash with a SEGV. To receive a patch, users should upgrade to one of the following versions: 18.24.3, 20.9.3, 21.4.3, certified-18.9-cert12, certified-20.7-cert2. Two workarounds are available. Disable res_resolver_unbound by setting `noload = res_resolver_unbound.so` in modules.conf, or set `rewrite_contact = yes` on all PJSIP endpoints. NOTE: This may not be appropriate for all Asterisk configurations. |
| 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. |
| Asterisk is an open-source private branch exchange (PBX). Prior to versions 18.26.2, 20.14.1, 21.9.1, and 22.4.1 of Asterisk and versions 18.9-cert14 and 20.7-cert5 of certified-asterisk, trying to disallow shell commands to be run via the Asterisk command line interface (CLI) by configuring `cli_permissions.conf` (e.g. with the config line `deny=!*`) does not work which could lead to a security risk. If an administrator running an Asterisk instance relies on the `cli_permissions.conf` file to work and expects it to deny all attempts to execute shell commands, then this could lead to a security vulnerability. Versions 18.26.2, 20.14.1, 21.9.1, and 22.4.1 of Asterisk and versions 18.9-cert14 and 20.7-cert5 of certified-asterisk fix the issue. |
| Asterisk is an open-source private branch exchange (PBX). Prior to versions 18.26.2, 20.14.1, 21.9.1, and 22.4.1 of Asterisk and versions 18.9-cert14 and 20.7-cert5 of certified-asterisk, SIP requests of the type MESSAGE (RFC 3428) authentication do not get proper alignment. An authenticated attacker can spoof any user identity to send spam messages to the user with their authorization token. Abuse of this security issue allows authenticated attackers to send fake chat messages can be spoofed to appear to come from trusted entities. Even administrators who follow Security best practices and Security Considerations can be impacted. Therefore, abuse can lead to spam and enable social engineering, phishing and similar attacks. Versions 18.26.2, 20.14.1, 21.9.1, and 22.4.1 of Asterisk and versions 18.9-cert14 and 20.7-cert5 of certified-asterisk fix the issue. |
| Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 18.26.4 and 18.9-cert17, RTP UDP ports and internal resources can leak due to a lack of session termination. This could result in leaks and resource exhaustion. This issue has been patched in versions 18.26.4 and 18.9-cert17. |
| Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.15.2, 21.10.2, and 22.5.2, if a SIP request is received with an Authorization header that contains a realm that wasn't in a previous 401 response's WWW-Authenticate header, or an Authorization header with an incorrect realm was received without a previous 401 response being sent, the get_authorization_header() function in res_pjsip_authenticator_digest will return a NULL. This wasn't being checked before attempting to get the digest algorithm from the header which causes a SEGV. This issue has been patched in versions 20.15.2, 21.10.2, and 22.5.2. There are no workarounds. |
| Asterisk is an open source private branch exchange and telephony toolkit. After upgrade to 18.23.0, ALL unauthorized SIP requests are identified as PJSIP Endpoint of local asterisk server. This vulnerability is fixed in 18.23.1, 20.8.1, and 21.3.1.
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