Asterisk

In Asterisk 11.x before 11.25.2, 13.x before 13.17.1, and 14.x before 14.6.1 and Certified Asterisk 11.x before 11.6-cert17 and 13.x before 13.13-cert5, unauthorized command execution is possible. The app_minivm module has an "externnotify" program configuration option that is…

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…

A Buffer Overflow issue was discovered in Asterisk Open Source 13 before 13.18.1, 14 before 14.7.1, and 15 before 15.1.1 and Certified Asterisk 13.13 before 13.13-cert7. No size checking is done when setting the user field for Party B on a CDR. Thus, it is possible for someone…

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…

An issue was discovered in Asterisk 13.18.4 and older, 14.7.4 and older, 15.1.4 and older, and 13.18-cert1 and older. A select set of SIP messages create a dialog in Asterisk. Those SIP messages must contain a contact header. For those messages, if the header was not present and…

There is a stack consumption vulnerability in the res_http_websocket.so module of Asterisk through 13.23.0, 14.7.x through 14.7.7, and 15.x through 15.6.0 and Certified Asterisk through 13.21-cert2. It allows an attacker to crash Asterisk via a specially crafted HTTP request to…

In the pjsip channel driver (res_pjsip) in Asterisk 13.x before 13.17.1 and 14.x before 14.6.1, a carefully crafted tel URI in a From, To, or Contact header could cause Asterisk to crash.

A NULL pointer access issue was discovered in Asterisk 15.x through 15.2.1. The RTP support in Asterisk maintains its own registry of dynamic codecs and desired payload numbers. While an SDP negotiation may result in a codec using a different payload number, these desired ones…

In Asterisk 11.x before 11.25.3, 13.x before 13.17.2, and 14.x before 14.6.2 and Certified Asterisk 11.x before 11.6-cert18 and 13.x before 13.13-cert6, insufficient RTCP packet validation could allow reading stale buffer contents and when combined with the "nat" and…

In res/res_rtp_asterisk.c in Asterisk 11.x before 11.25.2, 13.x before 13.17.1, and 14.x before 14.6.1 and Certified Asterisk 11.x before 11.6-cert17 and 13.x before 13.13-cert5, unauthorized data disclosure (media takeover in the RTP stack) is possible with careful timing by an…

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…

chain_sip in Asterisk Open Source 11.x before 11.23.1 and 13.x 13.11.1 and Certified Asterisk 11.6 before 11.6-cert15 and 13.8 before 13.8-cert3 allows remote attackers to cause a denial of service (port exhaustion).

An issue was discovered in Asterisk Open Source 13.12.x and 13.13.x before 13.13.1 and 14.x before 14.2.1. If an SDP offer or answer is received with the Opus codec and with the format parameters separated using a space the code responsible for parsing will recursively call…

The IAX2 channel driver (chan_iax2) in Asterisk Open 1.2.x before 1.2.23, 1.4.x before 1.4.9, and Asterisk Appliance Developer Kit before 0.6.0, when configured to allow unauthenticated calls, allows remote attackers to cause a denial of service (resource exhaustion) via a flood…

An 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. res_pjsip allows remote authenticated users to crash Asterisk (segmentation fault) by sending a number of SIP INVITE messages on a TCP…

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…

Asterisk Open Source 1.8.x, 11.x before 11.21.1, 12.x, and 13.x before 13.7.1 and Certified Asterisk 1.8.28, 11.6 before 11.6-cert12, and 13.1 before 13.1-cert3 allow remote authenticated users to cause a denial of service (uninitialized pointer dereference and crash) via a zero…

A Remote Crash issue was discovered in Asterisk Open Source 13.x before 13.18.4, 14.x before 14.7.4, and 15.x before 15.1.4 and Certified Asterisk before 13.13-cert9. Certain compound RTCP packets cause a crash in the RTCP Stack.

An issue was discovered in Asterisk Open Source 13 before 13.18.1, 14 before 14.7.1, and 15 before 15.1.1 and Certified Asterisk 13.13 before 13.13-cert7. A memory leak occurs when an Asterisk pjsip session object is created and that call gets rejected before the session itself…

chan_sip in Asterisk Open Source 1.8.x, 11.x before 11.21.1, 12.x, and 13.x before 13.7.1 and Certified Asterisk 1.8.28, 11.6 before 11.6-cert12, and 13.1 before 13.1-cert3, when the timert1 sip.conf configuration is set to a value greater than 1245, allows remote attackers to…

An issue was discovered in Asterisk Open Source 13.x before 13.21.1, 14.x before 14.7.7, and 15.x before 15.4.1 and Certified Asterisk 13.18-cert before 13.18-cert4 and 13.21-cert before 13.21-cert2. When endpoint specific ACL rules block a SIP request, they respond with a 403…

An issue was discovered in Asterisk Open Source 11.x before 11.25.1, 13.x before 13.13.1, and 14.x before 14.2.1 and Certified Asterisk 11.x before 11.6-cert16 and 13.x before 13.8-cert4. The chan_sip channel driver has a liberal definition for whitespace when attempting to…

The Skinny channel driver (chan_skinny) in Asterisk before 1.2.22 and 1.4.x before 1.4.8, Business Edition before B.2.2.1, AsteriskNOW before beta7, Appliance Developer Kit before 0.5.0, and s800i before 1.0.2 allows remote attackers to cause a denial of service (crash) via a…

The IAX2 protocol implementation in Asterisk Open Source 1.0.x, 1.2.x before 1.2.30, and 1.4.x before 1.4.21.2; Business Edition A.x.x, B.x.x before B.2.5.4, and C.x.x before C.1.10.3; AsteriskNOW; Appliance Developer Kit 0.x.x; and s800i 1.0.x before 1.2.0.1 allows remote…

The SIP channel driver in Asterisk Open Source 1.4.x before 1.4.17, Business Edition before C.1.0-beta8, AsteriskNOW before beta7, Appliance Developer Kit before Asterisk 1.4 revision 95946, and Appliance s800i 1.0.x before 1.0.3.4 allows remote attackers to cause a denial of…

The IAX2 channel driver (chan_iax2) in Asterisk before 1.2.22 and 1.4.x before 1.4.8, Business Edition before B.2.2.1, AsteriskNOW before beta7, Appliance Developer Kit before 0.5.0, and s800i before 1.0.2 allows remote attackers to cause a denial of service (crash) via 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…

Asterisk 1.4 before 1.4.1 and 1.2 before 1.2.16 allows remote attackers to cause a denial of service (crash) by sending a Session Initiation Protocol (SIP) packet without a URI and SIP-version header, which results in a NULL pointer dereference.

Directory traversal vulnerability in vmail.cgi in Asterisk 1.0.9 through 1.2.0-beta1 allows remote attackers to access WAV files via a .. (dot dot) in the folder parameter.

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…

Asterisk Open Source 1.8 before 1.8.32.3, 11.x before 11.17.1, 12.x before 12.8.2, and 13.x before 13.3.2 and Certified Asterisk 1.8.28 before 1.8.28-cert5, 11.6 before 11.6-cert11, and 13.1 before 13.1-cert2, when registering a SIP TLS device, does not properly handle a null…

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…

Asterisk Open Source 1.0.x and 1.2.x before 1.2.29 and Business Edition A.x.x and B.x.x before B.2.5.3, when pedantic parsing (aka pedanticsipchecking) is enabled, allows remote attackers to cause a denial of service (daemon crash) via a SIP INVITE message that lacks a From…

Multiple buffer overflows in Asterisk Open Source 1.4.x before 1.4.18.1 and 1.4.19-rc3, Open Source 1.6.x before 1.6.0-beta6, Business Edition C.x.x before C.1.6.1, AsteriskNOW 1.0.x before 1.0.2, Appliance Developer Kit before 1.4 revision 109386, and s800i 1.1.x before 1.1.0.2…

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.

Asterisk is an open source private branch exchange (PBX) and telephony toolkit. Prior to asterisk versions 18.24.2, 20.9.2, and 21.4.2 and certified-asterisk versions 18.9-cert11 and 20.7-cert2, an AMI user with `write=originate` may change all configuration files in the…

Multiple SQL injection vulnerabilities in cdr_addon_mysql in Asterisk-Addons before 1.2.8, and 1.4.x before 1.4.4, allow remote attackers to execute arbitrary SQL commands via the (1) source and (2) destination numbers, and probably (3) SIP URI, when inserting a record.

Double free vulnerability in the WebSocket Server (res_http_websocket module) in Asterisk Open Source 11.x before 11.14.2, 12.x before 12.7.2, and 13.x before 13.0.2 and Certified Asterisk 11.6 before 11.6-cert9 allows remote attackers to cause a denial of service (crash) by…

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…

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…

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…

Asterisk 1.2.10 supports the use of client-controlled variables to determine filenames in the Record function, which allows remote attackers to (1) execute code via format string specifiers or (2) overwrite files via directory traversals involving unspecified vectors, as…

Stack-based buffer overflow in channels/chan_mgcp.c in MGCP in Asterisk 1.0 through 1.2.10 allows remote attackers to execute arbitrary code via a crafted audit endpoint (AUEP) response.

Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, the asterisk/contrib/scripts/ast_coredumper runs as root, as noted by the NOTES tag on line 689 of the ast_coredumper file. The script…

Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, when ast_coredumper writes its gdb init and output files to a directory that is world-writable (for example /tmp), an attacker with write…

Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, the ast_xml_open() function in xml.c parses XML documents using libxml with unsafe parsing options that enable entity expansion and…

Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, user supplied/control values for Cookies and any GET variable query Parameter are directly interpolated into the HTML of the page using…

A local privilege escalation vulnerability exists in the safe_asterisk script included with the Asterisk toolkit package. When Asterisk is started via this script (common in SysV init or FreePBX environments), it sources all .sh files located in /etc/asterisk/startup.d/ as root,…

Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 20.15.2, 21.10.2, and 22.5.2, if a SIP request is received with an Authorization header that contains a realm that wasn't in a previous 401 response's WWW-Authenticate header, or an…

Asterisk is an open source private branch exchange and telephony toolkit. Prior to versions 18.26.4 and 18.9-cert17, RTP UDP ports and internal resources can leak due to a lack of session termination. This could result in leaks and resource exhaustion. This issue has been…