asterisk

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 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.

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, 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 (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 `cli_permissions.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 `cli_permissions.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 (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.

In Sangoma Asterisk through 16.28.0, 17.x and 18.x through 18.14.0, and 19.x through 19.6.0, an incoming Setup message to addons/ooh323c/src/ooq931.c with a malformed Calling or Called Party IE can cause a crash.

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 get_authorization_header() function in res_pjsip_authenticator_digest 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 allows calls on prohibited networks

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.

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.

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/res_stir_shaken /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.

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.

A use-after-free in res_pjsip_pubsub.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.

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. 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 (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 res_resolver_unbound 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 res_resolver_unbound by setting `noload = res_resolver_unbound.so` in modules.conf, or set `rewrite_contact = yes` on all PJSIP endpoints. NOTE: This may not be appropriate for all Asterisk configurations.

An issue in the action_listcategories() 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 res_pjsip_session 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.

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.