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 was discovered in res_pjsip_diversion.c in Sangoma Asterisk before 13.38.0, 14.x through 16.x before 16.15.0, 17.x before 17.9.0, and 18.x before 18.1.0. A crash can occur when a SIP message is received with a History-Info header that contains a tel-uri, or when a SIP 181 response is received that contains a tel-uri in the Diversion header.

A Buffer Overflow issue was discovered in Asterisk through 13.19.1, 14.x through 14.7.5, and 15.x through 15.2.1, and Certified Asterisk through 13.18-cert2. When processing a SUBSCRIBE request, the res_pjsip_pubsub module stores the accepted formats present in the Accept headers of the request. This code did not limit the number of headers it processed, despite having a fixed limit of 32. If more than 32 Accept headers were present, the code would write outside of its memory and cause a crash.

An issue was discovered in chan_skinny.c in Asterisk Open Source 13.18.2 and older, 14.7.2 and older, and 15.1.2 and older, and Certified Asterisk 13.13-cert7 and older. If the chan_skinny (aka SCCP protocol) channel driver is flooded with certain requests, it can cause the asterisk process to use excessive amounts of virtual memory, eventually causing asterisk to stop processing requests of any kind.

The PJSIP Channel Driver in Asterisk Open Source before 12.3.1 allows remote attackers to cause a denial of service (deadlock) by terminating a subscription request before it is complete, which triggers a SIP transaction timeout.

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 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.

The SIP over UDP implementation in Asterisk Open Source 1.4.x before 1.4.43, 1.6.x before 1.6.2.21, and 1.8.x before 1.8.7.2 uses different port numbers for responses to invalid requests depending on whether a SIP username exists, which allows remote attackers to enumerate usernames via a series of requests.

The default configuration of the SIP channel driver in Asterisk Open Source 1.4.x through 1.4.41.2 and 1.6.2.x through 1.6.2.18.2 does not enable the alwaysauthreject option, which allows remote attackers to enumerate account names by making a series of invalid SIP requests and observing the differences in the responses for different usernames, a different vulnerability than CVE-2011-2536.

chan_sip.c in the SIP channel driver in Asterisk Open Source 1.4.x before 1.4.41.2, 1.6.2.x before 1.6.2.18.2, and 1.8.x before 1.8.4.4, and Asterisk Business Edition C.3.x before C.3.7.3, disregards the alwaysauthreject option and generates different responses for invalid SIP requests depending on whether the user account exists, which allows remote attackers to enumerate account names via a series of requests.

chan_iax2.c in the IAX2 channel driver in Asterisk Open Source 1.4.x before 1.4.41.1, 1.6.2.x before 1.6.2.18.1, and 1.8.x before 1.8.4.3, and Asterisk Business Edition C.3 before C.3.7.3, accesses a memory address contained in an option control frame, which allows remote attackers to cause a denial of service (daemon crash) or possibly have unspecified other impact via a crafted frame.

manager.c in the Manager Interface in Asterisk Open Source 1.4.x before 1.4.40.1, 1.6.1.x before 1.6.1.25, 1.6.2.x before 1.6.2.17.3, and 1.8.x before 1.8.3.3 and Asterisk Business Edition C.x.x before C.3.6.4 does not properly check for the system privilege, which allows remote authenticated users to execute arbitrary commands via an Originate action that has an Async header in conjunction with an Application header.

Asterisk Open Source 1.4.x before 1.4.40.1, 1.6.1.x before 1.6.1.25, 1.6.2.x before 1.6.2.17.3, and 1.8.x before 1.8.3.3 and Asterisk Business Edition C.x.x before C.3.6.4 do not restrict the number of unauthenticated sessions to certain interfaces, which allows remote attackers to cause a denial of service (file descriptor exhaustion and disk space exhaustion) via a series of TCP connections.

Multiple stack-based and heap-based buffer overflows in the (1) decode_open_type and (2) udptl_rx_packet functions in main/udptl.c in Asterisk Open Source 1.4.x before 1.4.39.2, 1.6.1.x before 1.6.1.22, 1.6.2.x before 1.6.2.16.2, and 1.8 before 1.8.2.4; Business Edition C.x.x before C.3.6.3; AsteriskNOW 1.5; and s800i (Asterisk Appliance), when T.38 support is enabled, allow remote attackers to cause a denial of service (crash) and possibly execute arbitrary code via a crafted UDPTL packet.

Stack-based buffer overflow in the ast_uri_encode function in main/utils.c in Asterisk Open Source before 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.1, 1.8.1.2, 1.8.2.; and Business Edition before C.3.6.2; when running in pedantic mode allows remote authenticated users to execute arbitrary code via crafted caller ID data in vectors involving the (1) SIP channel driver, (2) URIENCODE dialplan function, or (3) AGI dialplan function.

The design of the dialplan functionality in Asterisk Open Source 1.2.x, 1.4.x, and 1.6.x; and Asterisk Business Edition B.x.x and C.x.x, when using the ${EXTEN} channel variable and wildcard pattern matches, allows context-dependent attackers to inject strings into the dialplan using metacharacters that are injected when the variable is expanded, as demonstrated using the Dial application to process a crafted SIP INVITE message that adds an unintended outgoing channel leg. NOTE: it could be argued that this is not a vulnerability in Asterisk, but a class of vulnerabilities that can occur in any program that uses this feature without the associated filtering functionality that is already available.

rtp.c in Asterisk Open Source 1.2.x before 1.2.37, 1.4.x before 1.4.27.1, 1.6.0.x before 1.6.0.19, and 1.6.1.x before 1.6.1.11; Business Edition B.x.x before B.2.5.13, C.2.x.x before C.2.4.6, and C.3.x.x before C.3.2.3; and s800i 1.3.x before 1.3.0.6 allows remote attackers to cause a denial of service (daemon crash) via an RTP comfort noise payload with a long data length.