PatchFirst · Vulnerabilidades recientes
Vulnerabilidades publicadas para asterisk en los últimos 30 días: 4. Incluidas en el catálogo KEV: 0.
Vulnerabilidades: 213
ConfBridge in Asterisk 11.x before 11.14.1 and Certified Asterisk 11.6 before 11.6-cert8 does not properly handle state changes, which allows remote attackers to cause a denial of service (channel hang and memory consumption) by causing transitions to be delayed, which triggers a state change from hung up to waiting for media.
The res_pjsip_acl module in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1 does not properly create and load ACLs defined in pjsip.conf at startup, which allows remote attackers to bypass intended PJSIP ACL rules.
The (1) VoIP channel drivers, (2) DUNDi, and (3) Asterisk Manager Interface (AMI) in Asterisk Open Source 1.8.x before 1.8.32.1, 11.x before 11.14.1, 12.x before 12.7.1, and 13.x before 13.0.1 and Certified Asterisk 1.8.28 before 1.8.28-cert3 and 11.6 before 11.6-cert8 allows remote attackers to bypass the ACL restrictions via a packet with a source IP that does not share the address family as the first ACL entry.
Asterisk Open Source 1.8.x before 1.8.28.1, 11.x before 11.10.1, and 12.x before 12.3.1 and Certified Asterisk 1.8.15 before 1.8.15-cert6 and 11.6 before 11.6-cert3 allows remote attackers to cause a denial of service (connection consumption) via a large number of (1) inactive or (2) incomplete HTTP connections.
Asterisk Open Source 11.x before 11.10.1 and 12.x before 12.3.1 and Certified Asterisk 11.6 before 11.6-cert3 allows remote authenticated Manager users to execute arbitrary shell commands via a MixMonitor action.
channels/chan_sip.c in Asterisk Open Source 1.8.x before 1.8.26.1, 11.8.x before 11.8.1, and 12.1.x before 12.1.1, and Certified Asterisk 1.8.15 before 1.8.15-cert5 and 11.6 before 11.6-cert2, when chan_sip has a certain configuration, allows remote authenticated users to cause a denial of service (channel and file descriptor consumption) via an INVITE request with a (1) Session-Expires or (2) Min-SE header with a malformed or invalid value.
main/http.c in Asterisk Open Source 1.8.x before 1.8.26.1, 11.8.x before 11.8.1, and 12.1.x before 12.1.1, and Certified Asterisk 1.8.x before 1.8.15-cert5 and 11.6 before 11.6-cert2, allows remote attackers to cause a denial of service (stack consumption) and possibly execute arbitrary code via an HTTP request with a large number of Cookie headers.
Buffer overflow in the unpacksms16 function in apps/app_sms.c in Asterisk Open Source 1.8.x before 1.8.24.1, 10.x before 10.12.4, and 11.x before 11.6.1; Asterisk with Digiumphones 10.x-digiumphones before 10.12.4-digiumphones; and Certified Asterisk 1.8.x before 1.8.15-cert4 and 11.x before 11.2-cert3 allows remote attackers to cause a denial of service (daemon crash) via a 16-bit SMS message with an odd number of bytes, which triggers an infinite loop.
The SIP channel driver (channels/chan_sip.c) in Asterisk Open Source 1.8.x before 1.8.23.1, 10.x before 10.12.3, and 11.x before 11.5.1; Certified Asterisk 1.8.15 before 1.8.15-cert3 and 11.2 before 11.2-cert2; and Asterisk Digiumphones 10.x-digiumphones before 10.12.3-digiumphones allows remote attackers to cause a denial of service (NULL pointer dereference, segmentation fault, and daemon crash) via an invalid SDP that defines a media description before the connection description in a SIP request.
The SIP channel driver (channels/chan_sip.c) in Asterisk Open Source 1.8.17.x through 1.8.22.x, 1.8.23.x before 1.8.23.1, and 11.x before 11.5.1 and Certified Asterisk 1.8.15 before 1.8.15-cert3 and 11.2 before 11.2-cert2 allows remote attackers to cause a denial of service (NULL pointer dereference, segmentation fault, and daemon crash) via an ACK with SDP to a previously terminated channel. NOTE: some of these details are obtained from third party information.
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.
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.
Asterisk Open Source 1.8.x before 1.8.19.1, 10.x before 10.11.1, and 11.x before 11.1.2; Certified Asterisk 1.8.11 before 1.8.11-cert10; and Asterisk Digiumphones 10.x-digiumphones before 10.11.1-digiumphones, when anonymous calls are enabled, allow remote attackers to cause a denial of service (resource consumption) by making anonymous calls from multiple sources and consequently adding many entries to the device state cache.
Multiple stack consumption vulnerabilities in Asterisk Open Source 1.8.x before 1.8.19.1, 10.x before 10.11.1, and 11.x before 11.1.2; Certified Asterisk 1.8.11 before 1.8.11-cert10; and Asterisk Digiumphones 10.x-digiumphones before 10.11.1-digiumphones allow remote attackers to cause a denial of service (daemon crash) via TCP data using the (1) SIP, (2) HTTP, or (3) XMPP protocol.
Stack-based buffer overflow in the ast_parse_digest function in main/utils.c in Asterisk 1.8.x before 1.8.10.1 and 10.x before 10.2.1 allows remote attackers to cause a denial of service (crash) or possibly execute arbitrary code via a long string in an HTTP Digest Authentication header.
Stack-based buffer overflow in the milliwatt_generate function in the Miliwatt application in Asterisk 1.4.x before 1.4.44, 1.6.x before 1.6.2.23, 1.8.x before 1.8.10.1, and 10.x before 10.2.1, when the o option is used and the internal_timing option is off, allows remote attackers to cause a denial of service (application crash) via a large number of samples in an audio packet.
channels/chan_iax2.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-cert7, Asterisk Digiumphones 10.x.x-digiumphones before 10.7.1-digiumphones, and Asterisk Business Edition C.3.x before C.3.7.6 does not enforce ACL rules during certain uses of peer credentials, which allows remote authenticated users to bypass intended outbound-call restrictions by leveraging the availability of these credentials.
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.
Double free vulnerability in apps/app_voicemail.c in Asterisk Open Source 1.8.x before 1.8.13.1 and 10.x before 10.5.2, Certified Asterisk 1.8.11-certx before 1.8.11-cert4, and Asterisk Digiumphones 10.x.x-digiumphones before 10.5.2-digiumphones allows remote authenticated users to cause a denial of service (daemon crash) by establishing multiple voicemail sessions and accessing both the Urgent mailbox and the INBOX mailbox.
channels/chan_sip.c in Asterisk Open Source 1.8.x before 1.8.13.1 and 10.x before 10.5.2, Asterisk Business Edition C.3.x before C.3.7.5, Certified Asterisk 1.8.11-certx before 1.8.11-cert4, and Asterisk Digiumphones 10.x.x-digiumphones before 10.5.2-digiumphones does not properly handle a provisional response to a SIP reINVITE request, which allows remote authenticated users to cause a denial of service (RTP port exhaustion) via sessions that lack final responses.
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