IAX2 in Asterisk Open Source 1.2.x before 1.2.31, 1.4.x before 1.4.23-rc4, and 1.6.x before 1.6.0.3-rc2; Business Edition A.x.x, B.x.x before B.2.5.7, C.1.x.x before C.1.10.4, and C.2.x.x before C.2.1.2.1; and s800i 1.2.x before 1.3.0 responds differently to a failed login attempt depending on whether the user account exists, which allows remote attackers to enumerate valid usernames.

Array index error in the dahdi/tor2.c driver in Zaptel (aka DAHDI) 1.4.11 and earlier allows local users in the dialout group to overwrite an integer value in kernel memory by writing to /dev/zap/ctl, related to an incorrect tor2 patch for CVE-2008-5396 that uses the wrong variable in a range check against the value of lc->sync.

Asterisk Open Source 1.2.26 through 1.2.30.3 and Business Edition B.2.3.5 through B.2.5.5, when realtime IAX2 users are enabled, allows remote attackers to cause a denial of service (crash) via authentication attempts involving (1) an unknown user or (2) a user using hostname matching.

Asterisk Open Source 1.6.0.x before 1.6.0.22, 1.6.1.x before 1.6.1.14, and 1.6.2.x before 1.6.2.2, and Business Edition C.3 before C.3.3.2, allows remote attackers to cause a denial of service (daemon crash) via an SIP T.38 negotiation with an SDP FaxMaxDatagram field that is (1) missing, (2) modified to contain a negative number, or (3) modified to contain a large number.

A local privilege escalation vulnerability exists in the safeasterisk 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, without validating ownership or permissions.

Non-root users with legitimate write access to /etc/asterisk can exploit this behaviour by placing malicious scripts in the startup.d directory, which will then execute with root privileges upon service restart.

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 aststrappend. The endpoint at GET /httpstatus is the potential vulnerable endpoint relating to asterisk/main /http.c. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2.

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 astxmlopen() function in xml.c parses XML documents using libxml with unsafe parsing options that enable entity expansion and XInclude processing. Specifically, it invokes xmlReadFile() with the XMLPARSENOENT flag and later processes XIncludes via xmlXIncludeProcess().If any untrusted or user-supplied XML file is passed to this function, it can allow an attacker to trigger XML External Entity (XXE) or XInclude-based local file disclosure, potentially exposing sensitive files from the host system. This can also be triggered in other cases in which the user is able to supply input in xml format that triggers the asterisk process to parse it. This issue has been patched in versions 20.7-cert9, 20.18.2, 21.12.1, 22.8.2, and 23.2.2.

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.

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.

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) 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 (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. 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 patched in versions 18.26.4 and 18.9-cert17.

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. 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.

Asterisk is an open source private branch exchange and telephony toolkit. In versions up to and including 18.26.2, between 20.00.0 and 20.15.0, 20.7-cert6, 21.00.0, 22.00.0 through 22.5.0, there is a remote DoS and possible RCE condition in asterisk/res/resstirshaken /verification.c that can be exploited when an attacker can set an arbitrary Identity header, or STIR/SHAKEN is enabled, with verification set in the SIP profile associated with the endpoint to be attacked. This is fixed in versions 18.26.3, 20.7-cert6, 20.15.1, 21.10.1 and 22.5.1.

A denial of service flaw was found in the way asterisk processed certain requests to negotiate secure video stream, when the ressrtp Asterisk module has been loaded and video support has not been enabled. A remote attacker could provide a specially-crafted media stream negotiation request, which once processed by Asterisk would lead to asterisk daemon crash by processing crypto line for such media stream.

References: [1] http://downloads.asterisk.org/pub/security/AST-2012-001.html [2] https://issues.asterisk.org/jira/browse/ASTERISK-19202

Upstream patch against the v1.8.x branch: [3] http://downloads.asterisk.org/pub/security/AST-2012-001-1.8.diff

Upstream patch against the v1.10.x branch: [4] http://downloads.asterisk.org/pub/security/AST-2012-001-10.diff

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.

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.

chanskinny.c in the Skinny (aka SCCP) channel driver in Certified Asterisk 1.8.11-cert before 1.8.11-cert2 and Asterisk Open Source 1.8.x before 1.8.12.1 and 10.x before 10.4.1 allows remote authenticated users to cause a denial of service (NULL pointer dereference and daemon crash) by closing a connection in off-hook mode.

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.