CVE |
Vendors |
Products |
Updated |
CVSS v3.1 |
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 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 (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. |
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.
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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. |
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 `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. |
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 memory exhaustion vulnerability exists in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1 and Certified Asterisk 13.13 before 13.13-cert4, which can be triggered by sending specially crafted SCCP packets causing an infinite loop and leading to memory exhaustion (by message logging in that loop). |
chan_skinny.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. |
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. |
Multiple SQL injection vulnerabilities in cdr_addon_mysql in Asterisk-Addons before 1.2.8, and 1.4.x before 1.4.4, allow remote attackers to execute arbitrary SQL commands via the (1) source and (2) destination numbers, and probably (3) SIP URI, when inserting a record. |
The IAX2 channel driver (chan_iax2) 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 cause a denial of service (crash) via a crafted (1) LAGRQ or (2) LAGRP frame that contains information elements of IAX frames, which results in a NULL pointer dereference when Asterisk does not properly set an associated variable. |
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 SIP channel driver in Asterisk Open Source 1.4.x before 1.4.17, Business Edition before C.1.0-beta8, AsteriskNOW before beta7, Appliance Developer Kit before Asterisk 1.4 revision 95946, and Appliance s800i 1.0.x before 1.0.3.4 allows remote attackers to cause a denial of service (daemon crash) via a BYE message with an Also (Also transfer) header, which triggers a NULL pointer dereference. |
Stack-based buffer overflow in the IAX2 channel driver (chan_iax2) 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 IAX2 channel driver (chan_iax2) in Asterisk 1.2 before revision 72630 and 1.4 before revision 65679, when configured to allow unauthenticated calls, sends "early audio" to an unverified source IP address of a NEW message, which allows remote attackers to cause a denial of service (traffic amplification) via a spoofed NEW message. |
Asterisk Open Source 1.2.x before 1.2.26 and 1.4.x before 1.4.16, and Business Edition B.x.x before B.2.3.6 and C.x.x before C.1.0-beta8, when using database-based registrations ("realtime") and host-based authentication, does not check the IP address when the username is correct and there is no password, which allows remote attackers to bypass authentication using a valid username. |