An issue was discovered in Asterisk Open Source 13.x before 13.21.1, 14.x before 14.7.7, and 15.x before 15.4.1 and Certified Asterisk 13.18-cert before 13.18-cert4 and 13.21-cert before 13.21-cert2. When endpoint specific ACL rules block a SIP request, they respond with a 403 forbidden. However, if an endpoint is not identified, then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints.

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.

An issue was discovered in Asterisk through 13.19.1, 14.x through 14.7.5, and 15.x through 15.2.1, and Certified Asterisk through 13.18-cert2. res_pjsip allows remote authenticated users to crash Asterisk (segmentation fault) by sending a number of SIP INVITE messages on a TCP or TLS connection and then suddenly closing the connection.

A Buffer Overflow issue was discovered in Asterisk through 13.19.1, 14.x through 14.7.5, and 15.x through 15.2.1, and Certified Asterisk through 13.18-cert2. When processing a SUBSCRIBE request, the res_pjsip_pubsub module stores the accepted formats present in the Accept headers of the request. This code did not limit the number of headers it processed, despite having a fixed limit of 32. If more than 32 Accept headers were present, the code would write outside of its memory and cause a crash.

An issue was discovered in res_http_websocket.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.

An issue was discovered in Asterisk 13.18.4 and older, 14.7.4 and older, 15.1.4 and older, and 13.18-cert1 and older. A select set of SIP messages create a dialog in Asterisk. Those SIP messages must contain a contact header. For those messages, if the header was not present and the PJSIP channel driver was used, Asterisk would crash. The severity of this vulnerability is somewhat mitigated if authentication is enabled. If authentication is enabled, a user would have to first be authorized before reaching the crash point.

A Remote Crash issue was discovered in Asterisk Open Source 13.x before 13.18.4, 14.x before 14.7.4, and 15.x before 15.1.4 and Certified Asterisk before 13.13-cert9. Certain compound RTCP packets cause a crash in the RTCP Stack.

An issue was discovered in chan_skinny.c in Asterisk Open Source 13.18.2 and older, 14.7.2 and older, and 15.1.2 and older, and Certified Asterisk 13.13-cert7 and older. If the chan_skinny (aka SCCP protocol) channel driver is flooded with certain requests, it can cause the asterisk process to use excessive amounts of virtual memory, eventually causing asterisk to stop processing requests of any kind.

An issue was discovered in Asterisk Open Source 13 before 13.18.1, 14 before 14.7.1, and 15 before 15.1.1 and Certified Asterisk 13.13 before 13.13-cert7. A memory leak occurs when an Asterisk pjsip session object is created and that call gets rejected before the session itself is fully established. When this happens the session object never gets destroyed. Eventually Asterisk can run out of memory and crash.