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Search Results (14 CVEs found)

CVE Vendors Products Updated CVSS v3.1
CVE-2024-42491 2 Asterisk, Sangoma 3 Asterisk, Asterisk, Certified Asterisk 2025-08-26 5.7 Medium
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.
CVE-2022-42706 1 Sangoma 2 Asterisk, Certified Asterisk 2025-04-24 4.9 Medium
An issue was discovered in Sangoma Asterisk through 16.28, 17 and 18 through 18.14, 19 through 19.6, and certified through 18.9-cert1. GetConfig, via Asterisk Manager Interface, allows a connected application to access files outside of the asterisk configuration directory, aka Directory Traversal.
CVE-2022-42705 1 Sangoma 2 Asterisk, Certified Asterisk 2025-04-24 6.5 Medium
A use-after-free in res_pjsip_pubsub.c in Sangoma Asterisk 16.28, 18.14, 19.6, and certified/18.9-cert2 may allow a remote authenticated attacker to crash Asterisk (denial of service) by performing activity on a subscription via a reliable transport at the same time that Asterisk is also performing activity on that subscription.
CVE-2022-21723 4 Asterisk, Debian, Sangoma and 1 more 4 Certified Asterisk, Debian Linux, Asterisk and 1 more 2025-04-23 9.1 Critical
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.
CVE-2022-23608 4 Asterisk, Debian, Sangoma and 1 more 4 Certified Asterisk, Debian Linux, Asterisk and 1 more 2025-04-23 8.1 High
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.
CVE-2017-9358 2 Asterisk, Sangoma 2 Certified Asterisk, Asterisk 2025-04-20 N/A
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).
CVE-2012-2186 2 Asterisk, Sangoma 5 Business Edition, Certified Asterisk, Digiumphones and 2 more 2025-04-11 N/A
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.
CVE-2012-2948 2 Asterisk, Sangoma 3 Certified Asterisk, Open Source, Asterisk 2025-04-11 N/A
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.
CVE-2023-49786 2 Digium, Sangoma 2 Asterisk, Certified Asterisk 2025-02-13 7.5 High
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.
CVE-2023-49294 2 Digium, Sangoma 2 Asterisk, Certified Asterisk 2025-02-13 4.9 Medium
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.
CVE-2023-37457 2 Digium, Sangoma 2 Asterisk, Certified Asterisk 2025-02-13 7.5 High
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.
CVE-2021-37706 4 Asterisk, Debian, Sangoma and 1 more 4 Certified Asterisk, Debian Linux, Asterisk and 1 more 2024-11-21 7.3 High
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.
CVE-2020-28327 2 Digium, Sangoma 2 Certified Asterisk, Asterisk 2024-11-21 5.3 Medium
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.
CVE-2020-28242 4 Asterisk, Debian, Fedoraproject and 1 more 4 Certified Asterisk, Debian Linux, Fedora and 1 more 2024-11-21 6.5 Medium
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.