The IAX2 channel driver (chaniax2) in Asterisk before 20070504 does not properly null terminate data, which allows remote attackers to trigger loss of transmitted data, and possibly obtain sensitive information (memory contents) or cause a denial of service (application crash), by sending a frame that lacks a 0 byte.

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 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.

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.

Stack-based buffer overflow in the IAX2 channel driver (chaniax2) 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 execute arbitrary code by sending a long (1) voice or (2) video RTP frame.

The AsteriskGUI HTTP server in Asterisk Open Source 1.4.x before 1.4.19-rc3 and 1.6.x before 1.6.0-beta6, Business Edition C.x.x before C.1.6, AsteriskNOW before 1.0.2, Appliance Developer Kit before revision 104704, and s800i 1.0.x before 1.1.0.2 generates insufficiently random manager ID values, which makes it easier for remote attackers to hijack a manager session via a series of ID guesses.

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.

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.

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 /etc/asterisk/ directory. This occurs because they are able to curl remote files and write them to disk, but are also able to append to existing files using the FILE function inside the SET application. This issue may result in privilege escalation, remote code execution and/or blind server-side request forgery with arbitrary protocol. Asterisk versions 18.24.2, 20.9.2, and 21.4.2 and certified-asterisk versions 18.9-cert11 and 20.7-cert2 contain a fix for this issue.

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/astcoredumper runs as root, as noted by the NOTES tag on line 689 of the astcoredumper file. The script will source the contents of /etc/asterisk/astdebugtools.conf, which resides in a folder that is writeable by the asterisk user:group. Due to the /etc/asterisk/astdebugtools.conf file following bash semantics and it being loaded; an attacker with write permissions may add or modify the file such that when the root astcoredumper is run; it would source and thereby execute arbitrary bash code found in the /etc/asterisk/astdebugtools.conf. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2.

Unspecified vulnerability in Asterisk Open Source 1.2.x before 1.2.27, 1.4.x before 1.4.18.1 and 1.4.19-rc3; Business Edition A.x.x, B.x.x before B.2.5.1, and C.x.x before C.1.6.2; AsteriskNOW 1.0.x before 1.0.2; Appliance Developer Kit before 1.4 revision 109393; and s800i 1.0.x before 1.1.0.2; allows remote attackers to access the SIP channel driver via a crafted From header.

Asterisk is an open source private branch exchange and telephony toolkit. Prior to Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3, an unbounded sscanf in respjsippubsub when parsing the MWI NOTIFY Message-Account field may allow an unauthenticated attacker to send a carefully crafted packet that causes a stack buffer overflow and corrupts the underlying transport, permanently disabling all SIP functionality. An Asterisk restart is required to restore full operation. This issue is fixed in Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3.

Asterisk is an open source private branch exchange and telephony toolkit. Prior to Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3, an authenticated Asterisk REST Interface (ARI) attacker with read-only privileges may use setChannelVar to execute write dialplan functions, including FILE() writes, even when livedangerously in asterisk.conf is set to no. The Asterisk HTTP web server must be enabled and reachable, and the attacker must already have valid read-only ARI credentials. HTTP is disabled by default and binds only to 127.0.0.1 by default even when enabled. This issue is fixed in Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3.

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. Prior to versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2, when astcoredumper writes its gdb init and output files to a directory that is world-writable (for example /tmp), an attacker with write permission(which is all users on a linux system) to that directory can cause root to execute arbitrary commands or overwrite arbitrary files by controlling the gdb init file and output paths. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2.

The SIP channel driver in Asterisk Open Source 1.2.x before 1.2.34, 1.4.x before 1.4.26.1, 1.6.0.x before 1.6.0.12, and 1.6.1.x before 1.6.1.4; Asterisk Business Edition A.x.x, B.x.x before B.2.5.9, C.2.x before C.2.4.1, and C.3.x before C.3.1; and Asterisk Appliance s800i 1.2.x before 1.3.0.3 does not use a maximum width when invoking sscanf style functions, which allows remote attackers to cause a denial of service (stack memory consumption) via SIP packets containing large sequences of ASCII decimal characters, as demonstrated via vectors related to (1) the CSeq value in a SIP header, (2) large Content-Length value, and (3) SDP.

The Manager Interface in Asterisk before 1.2.18 and 1.4.x before 1.4.3 allows remote attackers to cause a denial of service (crash) by using MD5 authentication to authenticate a user that does not have a password defined in manager.conf, resulting in a NULL pointer dereference.

The SIP channel driver (chansip) in Asterisk before 1.2.18 and 1.4.x before 1.4.3 does not properly parse SIP UDP packets that do not contain a valid response code, which allows remote attackers to cause a denial of service (crash).

The handleresponse function in chansip.c 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 Response code 0 in a SIP packet.

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.

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.

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 attackers to cause a denial of service (call-number exhaustion and CPU consumption) by quickly sending a large number of IAX2 (IAX) POKE requests.

The FWDOWNL firmware-download 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 attackers to cause a denial of service (traffic amplification) via an IAX2 FWDOWNL request.

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 and telephony toolkit. Prior to Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3, an unbounded memcpy of OGG/Speex packet data in formatoggspeex can cause a heap buffer overflow and crash Asterisk when reading a file containing corrupted Speex audio. The attacker must create the file and make it available to Asterisk either through sufficiently elevated privileges or by convincing an Asterisk administrator to make it available. Asterisk must also be configured to play that file. This issue is fixed in Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3.

Multiple stack-based buffer overflows in the processsdp function in chansip.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 demonstrated using SIP INVITE.

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 Authorization header with an incorrect realm was received without a previous 401 response being sent, the getauthorizationheader() function in respjsipauthenticatordigest will return a NULL. This wasn't being checked before attempting to get the digest algorithm from the header which causes a SEGV. This issue has been patched in versions 20.15.2, 21.10.2, and 22.5.2. There are no workarounds.

Asterisk is an open source private branch exchange and telephony toolkit. Prior to Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3, a NULL pointer dereference in the Asterisk Manager Interface (AMI) may allow an unauthenticated attacker to crash Asterisk when Digest Authentication is used via HTTP. The Asterisk HTTP web server, AMI, and AMI access via HTTP must all be enabled; each is disabled by default. The attacker must be able to connect to the HTTP server, which binds only to 127.0.0.1 by default even when enabled. This issue is fixed in Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 20.7-cert11, 22.8-cert3.

Asterisk is an open source private branch exchange and telephony toolkit. Prior to Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 22.8-cert3, an authenticated attacker with read-only permissions for the Asterisk REST Interface (ARI) may, under certain conditions, execute write operations through REST-over-WebSocket. The authorization bypass permits arbitrary module path loading and conditional remote code execution. The Asterisk HTTP web server must be enabled and reachable, and the attacker must already have valid read-only ARI credentials. HTTP is disabled by default and binds only to 127.0.0.1 by default even when enabled. This issue is fixed in Asterisk 20.20.1, 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 22.8-cert3.

Asterisk is an open source private branch exchange and telephony toolkit. From 21.0.0-pre1 until Asterisk 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 22.8-cert3, the chanooh323 Q.931 party-number parser lacks the CVE-2022-37325 fix because that fix was never forward-ported from Asterisk 20 to later versions. Malformed Q.931 elements in an OOH323 request can therefore crash Asterisk. This issue affects Asterisk versions greater than 20. The chanooh323 addon channel driver must be explicitly compiled and installed, which is not the default, and its port must be publicly reachable. This issue is fixed in Asterisk 21.12.3, 22.10.1, 23.4.1 and Certified Asterisk 22.8-cert3.