| CVE |
Vendors |
Products |
Updated |
CVSS v3.1 |
| 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. |
| The IAX2 protocol implementation in Asterisk Open Source 1.2.x before 1.2.35, 1.4.x before 1.4.26.2, 1.6.0.x before 1.6.0.15, and 1.6.1.x before 1.6.1.6; Business Edition B.x.x before B.2.5.10, C.2.x before C.2.4.3, and C.3.x before C.3.1.1; and s800i 1.3.x before 1.3.0.3 allows remote attackers to cause a denial of service (call-number exhaustion) by initiating many IAX2 message exchanges, a related issue to CVE-2008-3263. |
| 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 in the action_listcategories() function of Sangoma Asterisk v22/22.0.0/22.0.0-rc1/22.0.0-rc2/22.0.0-pre1 allows attackers to execute a path traversal. |
| 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 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. |
| An issue was discovered in Asterisk Open Source 15.x before 15.4.1. When connected to Asterisk via TCP/TLS, if the client abruptly disconnects, or sends a specially crafted message, then Asterisk gets caught in an infinite loop while trying to read the data stream. This renders the system unusable. |
| asterisk allows calls on prohibited networks |
