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.

Insecure Permissions vulnerability in asterisk v22 allows a remote attacker to execute arbitrary code via the actioncreateconfig function. NOTE: this is disputed by the Supplier because the impact is limited to creating empty files outside of the Asterisk product directory (aka directory traversal) and the attack can only be performed by a privileged user who has the ability to manage the configuration.

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.

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.

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.

Asterisk is an open-source private branch exchange (PBX). Prior to versions 18.24.3, 20.9.3, and 21.4.3 of Asterisk and versions 18.9-cert12 and 20.7-cert2 of certified-asterisk, if Asterisk attempts to send a SIP request to a URI whose host portion starts with .1 or [.1], and resresolverunbound is loaded, Asterisk will crash with a SEGV. To receive a patch, users should upgrade to one of the following versions: 18.24.3, 20.9.3, 21.4.3, certified-18.9-cert12, certified-20.7-cert2. Two workarounds are available. Disable resresolverunbound by setting noload = resresolverunbound.so in modules.conf, or set rewritecontact = yes on all PJSIP endpoints. NOTE: This may not be appropriate for all Asterisk configurations.

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.

Asterisk is an open source private branch exchange and telephony toolkit. After upgrade to 18.23.0, ALL unauthorized SIP requests are identified as PJSIP Endpoint of local asterisk server. This vulnerability is fixed in 18.23.1, 20.8.1, and 21.3.1.

An issue in the actionlistcategories() function of Sangoma Asterisk v22/22.0.0/22.0.0-rc1/22.0.0-rc2/22.0.0-pre1 allows attackers to execute a path traversal.

A use-after-free in respjsippubsub.c in Sangoma Asterisk 16.28, 18.14, 19.6, and certified/18.9-cert2 may allow a remote authenticated attacker to crash Asterisk (denial of service) by performing activity on a subscription via a reliable transport at the same time that Asterisk is also performing activity on that subscription.

An issue was discovered in Sangoma Asterisk through 16.28, 17 and 18 through 18.14, 19 through 19.6, and certified through 18.9-cert1. GetConfig, via Asterisk Manager Interface, allows a connected application to access files outside of the asterisk configuration directory, aka Directory Traversal.

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.