Filtered by vendor Asterisk
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Total
54 CVE
| CVE | Vendors | Products | Updated | CVSS v3.1 |
|---|---|---|---|---|
| CVE-2007-5690 | 1 Asterisk | 1 Zaptel | 2024-05-17 | N/A |
| Buffer overflow in sethdlc.c in the Asterisk Zaptel 1.4.5.1 might allow local users to gain privileges via a long device name (interface name) in the ifr_name field. NOTE: the vendor disputes this issue, stating that the application requires root access, so privilege boundaries are not crossed | ||||
| CVE-2020-28242 | 3 Asterisk, Debian, Fedoraproject | 4 Certified Asterisk, Open Source, Debian Linux and 1 more | 2023-11-07 | 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. | ||||
| CVE-2008-1897 | 1 Asterisk | 5 Asterisk Appliance Developer Kit, Asterisk Business Edition, Asterisknow and 2 more | 2023-10-20 | N/A |
| The IAX2 channel driver (chan_iax2) in Asterisk Open Source 1.0.x, 1.2.x before 1.2.28, and 1.4.x before 1.4.19.1; Business Edition A.x.x, B.x.x before B.2.5.2, and C.x.x before C.1.8.1; AsteriskNOW before 1.0.3; Appliance Developer Kit 0.x.x; and s800i before 1.1.0.3, when configured to allow unauthenticated calls, does not verify that an ACK response contains a call number matching the server's reply to a NEW message, which allows remote attackers to cause a denial of service (traffic amplification) via a spoofed ACK response that does not complete a 3-way handshake. NOTE: this issue exists because of an incomplete fix for CVE-2008-1923. | ||||
| CVE-2022-23608 | 4 Asterisk, Debian, Sangoma and 1 more | 4 Certified Asterisk, Debian Linux, Asterisk and 1 more | 2023-08-30 | 9.8 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 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-2022-21723 | 4 Asterisk, Debian, Sangoma and 1 more | 4 Certified Asterisk, Debian Linux, Asterisk and 1 more | 2023-08-30 | 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-2021-37706 | 4 Asterisk, Debian, Sangoma and 1 more | 4 Certified Asterisk, Debian Linux, Asterisk and 1 more | 2023-08-30 | 9.8 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 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-2021-46837 | 3 Asterisk, Debian, Digium | 3 Certified Asterisk, Debian Linux, Asterisk | 2023-01-28 | 6.5 Medium |
| 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. | ||||
| CVE-2012-0885 | 1 Asterisk | 1 Open Source | 2022-10-03 | N/A |
| chan_sip.c in Asterisk Open Source 1.8.x before 1.8.8.2 and 10.x before 10.0.1, when the res_srtp module is used and media support is improperly configured, allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a crafted SDP message with a crypto attribute and a (1) video or (2) text media type, as demonstrated by CSipSimple. | ||||
| CVE-2013-2685 | 1 Asterisk | 1 Open Source | 2022-10-03 | N/A |
| Stack-based buffer overflow in res/res_format_attr_h264.c in Asterisk Open Source 11.x before 11.2.2 allows remote attackers to execute arbitrary code via a long sprop-parameter-sets H.264 media attribute in a SIP Session Description Protocol (SDP) header. | ||||
| CVE-2013-2264 | 1 Asterisk | 4 Business Edition, Certified Asterisk, Digiumphones and 1 more | 2022-10-03 | N/A |
| The SIP channel driver in Asterisk Open Source 1.8.x before 1.8.20.2, 10.x before 10.12.2, and 11.x before 11.2.2; Certified Asterisk 1.8.15 before 1.8.15-cert2; Asterisk Business Edition (BE) C.3.x before C.3.8.1; and Asterisk Digiumphones 10.x-digiumphones before 10.12.2-digiumphones exhibits different behavior for invalid INVITE, SUBSCRIBE, and REGISTER transactions depending on whether the user account exists, which allows remote attackers to enumerate account names by (1) reading HTTP status codes, (2) reading additional text in a 403 (aka Forbidden) response, or (3) observing whether certain retransmissions occur. | ||||
| CVE-2013-2686 | 1 Asterisk | 3 Certified Asterisk, Digiumphones, Open Source | 2022-10-03 | N/A |
| main/http.c in the HTTP server in Asterisk Open Source 1.8.x before 1.8.20.2, 10.x before 10.12.2, and 11.x before 11.2.2; Certified Asterisk 1.8.15 before 1.8.15-cert2; and Asterisk Digiumphones 10.x-digiumphones before 10.12.2-digiumphones does not properly restrict Content-Length values, which allows remote attackers to conduct stack-consumption attacks and cause a denial of service (daemon crash) via a crafted HTTP POST request. NOTE: this vulnerability exists because of an incorrect fix for CVE-2012-5976. | ||||
| CVE-2020-28327 | 2 Asterisk, Digium | 2 Open Source, Certified Asterisk | 2020-11-20 | 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-2009-3723 | 2 Asterisk, Debian | 2 Open Source, Debian Linux | 2019-11-01 | 7.5 High |
| asterisk allows calls on prohibited networks | ||||
| CVE-2017-9358 | 1 Asterisk | 2 Certified Asterisk, Open Source | 2019-10-03 | 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-2018-12228 | 1 Asterisk | 1 Open Source | 2019-10-03 | N/A |
| 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. | ||||
| CVE-2007-2297 | 1 Asterisk | 1 Asterisk | 2018-10-16 | N/A |
| 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). | ||||
| CVE-2007-2294 | 1 Asterisk | 1 Asterisk | 2018-10-16 | N/A |
| 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. | ||||
| CVE-2007-2293 | 1 Asterisk | 1 Asterisk | 2018-10-16 | N/A |
| 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. | ||||
| CVE-2007-1594 | 1 Asterisk | 1 Asterisk | 2018-10-16 | N/A |
| 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. | ||||
| CVE-2007-1561 | 1 Asterisk | 1 Asterisk | 2018-10-16 | N/A |
| The channel driver 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 INVITE message with an SDP containing one valid and one invalid IP address. | ||||