main/translate.c in Sangoma Asterisk 13.28.0 and 16.5.0 allows a remote attacker to send a specific RTP packet during a call and cause a crash in a specific scenario.

An issue was discovered in reshttpwebsocket.c in Asterisk 15.x through 15.2.1. If the HTTP server is enabled (default is disabled), WebSocket payloads of size 0 are mishandled (with a busy loop).

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 are still stored internally. When an RTP packet was received, this registry would be consulted if the payload number was not found in the negotiated SDP. This registry was incorrectly consulted for all packets, even those which are dynamic. If the payload number resulted in a codec of a different type than the RTP stream (for example, the payload number resulted in a video codec but the stream carried audio), a crash could occur if no stream of that type had been negotiated. This was due to the code incorrectly assuming that a stream of that type would always exist.

Remote code execution can occur in Asterisk Open Source 13.x before 13.14.1 and 14.x before 14.3.1 and Certified Asterisk 13.13 before 13.13-cert3 because of a buffer overflow in a CDR user field, related to X-ClientCode in chansip, the CDR dialplan function, and the AMI Monitor action.

The PJSIP channel driver in Asterisk Open Source 12.x before 12.1.1, when qualifyfrequency "is enabled on an AOR and the remote SIP server challenges for authentication of the resulting OPTIONS request," allows remote attackers to cause a denial of service (crash) via a PJSIP endpoint that does not have an associated outgoing request.

res/respjsipextenstate.c in the PJSIP channel driver in Asterisk Open Source 12.x before 12.1.0 allows remote authenticated users to cause a denial of service (crash) via a SUBSCRIBE request without any Accept headers, which triggers an invalid pointer dereference.

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.

channels/chansip.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 chansip 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.

The SIP channel driver in Asterisk Open Source 1.4.22, 1.4.23, and 1.4.23.1; 1.6.0 before 1.6.0.6; 1.6.1 before 1.6.1.0-rc2; and Asterisk Business Edition C.2.3, with the pedantic option enabled, allows remote authenticated users to cause a denial of service (crash) via a SIP INVITE request without any headers, which triggers a NULL pointer dereference in the (1) sipuriheaderscmp and (2) sipuriparamscmp functions.

SQL injection vulnerability in the Call Detail Record (CDR) logging functionality for Asterisk allows remote attackers to execute arbitrary SQL via a CallerID string.

Buffer overflow in the getmsgtext of chansip.c in the Session Initiation Protocol (SIP) protocol implementation for Asterisk releases before August 15, 2003, allows remote attackers to execute arbitrary code via certain (1) MESSAGE or (2) INFO requests.

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.

main/acl.c in Asterisk Open Source 1.6.0.x before 1.6.0.25, 1.6.1.x before 1.6.1.17, and 1.6.2.x before 1.6.2.5 does not properly enforce remote host access controls when CIDR notation "/0" is used in permit= and deny= configuration rules, which causes an improper arithmetic shift and might allow remote attackers to bypass ACL rules and access services from unauthorized hosts.

Unspecified vulnerability in the SIP channel driver (channels/chansip.c) in Asterisk 1.2.x before 1.2.13 and 1.4.x before 1.4.0-beta3 allows remote attackers to cause a denial of service (resource consumption) via unspecified vectors that result in the creation of "a real pvt structure" that uses more resources than necessary.

Integer overflow in the getinput function in the Skinny channel driver (chanskinny.c) in Asterisk 1.0.x before 1.0.12 and 1.2.x before 1.2.13, as used by Cisco SCCP phones, allows remote attackers to execute arbitrary code via a certain dlen value that passes a signed integer comparison and leads to a heap-based buffer overflow.

Asterisk is an open-source private branch exchange (PBX). Prior to versions 18.26.2, 20.14.1, 21.9.1, and 22.4.1 of Asterisk and versions 18.9-cert14 and 20.7-cert5 of certified-asterisk, SIP requests of the type MESSAGE (RFC 3428) authentication do not get proper alignment. An authenticated attacker can spoof any user identity to send spam messages to the user with their authorization token. Abuse of this security issue allows authenticated attackers to send fake chat messages can be spoofed to appear to come from trusted entities. Even administrators who follow Security best practices and Security Considerations can be impacted. Therefore, abuse can lead to spam and enable social engineering, phishing and similar attacks. Versions 18.26.2, 20.14.1, 21.9.1, and 22.4.1 of Asterisk and versions 18.9-cert14 and 20.7-cert5 of certified-asterisk fix the issue.

Asterisk is an open-source private branch exchange (PBX). Prior to versions 18.26.2, 20.14.1, 21.9.1, and 22.4.1 of Asterisk and versions 18.9-cert14 and 20.7-cert5 of certified-asterisk, trying to disallow shell commands to be run via the Asterisk command line interface (CLI) by configuring clipermissions.conf (e.g. with the config line deny=!) does not work which could lead to a security risk. If an administrator running an Asterisk instance relies on the clipermissions.conf file to work and expects it to deny all attempts to execute shell commands, then this could lead to a security vulnerability. Versions 18.26.2, 20.14.1, 21.9.1, and 22.4.1 of Asterisk and versions 18.9-cert14 and 20.7-cert5 of certified-asterisk fix the issue.

An Integer Signedness issue (for a return code) in the respjsipsdprtp module in Digium Asterisk versions 15.7.1 and earlier and 16.1.1 and earlier allows remote authenticated users to crash Asterisk via a specially crafted SDP protocol violation.

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.

chanskinny.c in the Skinny (aka SCCP) channel driver in Asterisk Open Source 10.x before 10.5.1 allows remote authenticated users to cause a denial of service (NULL pointer dereference and daemon crash) by sending a Station Key Pad Button message and closing a connection in off-hook mode, a related issue to CVE-2012-2948.

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.

Buffer overflow in DNS SRV and NAPTR lookups in Digium Asterisk 15.x before 15.6.2 and 16.x before 16.0.1 allows remote attackers to crash Asterisk via a specially crafted DNS SRV or NAPTR response, because a buffer size is supposed to match an expanded length but actually matches a compressed length.

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.

Stack-based buffer overflow in res/resformatattrh264.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.

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.

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.

A respjsipsession 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.

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.

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.

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.