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 res_pjsip_t38.c in Sangoma Asterisk through 13.x and Certified Asterisk through 13.21-x. If it receives a re-invite initiating T.38 faxing and has a port of 0 and no c line in the SDP, a NULL pointer dereference and crash will occur. This is different from CVE-2019-18940.

reqresp_parser.c in the SIP channel driver in Asterisk Open Source 1.8.x before 1.8.4.3 allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a SIP packet with a Contact header that lacks a < (less than) character.

tcptls.c in the TCP/TLS server in Asterisk Open Source 1.6.1.x before 1.6.1.23, 1.6.2.x before 1.6.2.17.1, and 1.8.x before 1.8.3.1 allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) by establishing many short TCP sessions to services that use a certain TLS API.

manager.c in Asterisk Open Source 1.6.1.x before 1.6.1.24, 1.6.2.x before 1.6.2.17.2, and 1.8.x before 1.8.3.2 allows remote attackers to cause a denial of service (CPU and memory consumption) via a series of manager sessions involving invalid data.

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.

rtp.c in Asterisk Open Source 1.2.x before 1.2.37, 1.4.x before 1.4.27.1, 1.6.0.x before 1.6.0.19, and 1.6.1.x before 1.6.1.11; Business Edition B.x.x before B.2.5.13, C.2.x.x before C.2.4.6, and C.3.x.x before C.3.2.3; and s800i 1.3.x before 1.3.0.6 allows remote attackers to cause a denial of service (daemon crash) via an RTP comfort noise payload with a long data length.

main/rtp.c in Asterisk Open Source 1.6.1 before 1.6.1.2 allows remote attackers to cause a denial of service (crash) via an RTP text frame without a certain delimiter, which triggers a NULL pointer dereference and the subsequent calculation of an invalid pointer.

An issue was discovered in Sangoma Asterisk 13.x before 13.38.3, 16.x before 16.19.1, 17.x before 17.9.4, and 18.x before 18.5.1, and Certified Asterisk before 16.8-cert10. If the IAX2 channel driver receives a packet that contains an unsupported media format, a crash can occur.

Buffer overflow in DNS SRV and NAPTR lookups in Digium Asterisk 15.x before 15.6.2 and 16.x before 16.0.1 allows remote attackers to crash Asterisk via a specially crafted DNS SRV or NAPTR response, because a buffer size is supposed to match an expanded length but actually matches a compressed length.

Double free vulnerability in the WebSocket Server (res_http_websocket module) in Asterisk Open Source 11.x before 11.14.2, 12.x before 12.7.2, and 13.x before 13.0.2 and Certified Asterisk 11.6 before 11.6-cert9 allows remote attackers to cause a denial of service (crash) by sending a zero length frame after a non-zero length frame.

There is a stack consumption vulnerability in the res_http_websocket.so module of Asterisk through 13.23.0, 14.7.x through 14.7.7, and 15.x through 15.6.0 and Certified Asterisk through 13.21-cert2. It allows an attacker to crash Asterisk via a specially crafted HTTP request to upgrade the connection to a websocket.

ConfBridge in Asterisk 11.x before 11.14.1 and Certified Asterisk 11.6 before 11.6-cert8 does not properly handle state changes, which allows remote attackers to cause a denial of service (channel hang and memory consumption) by causing transitions to be delayed, which triggers a state change from hung up to waiting for media.

Use-after-free vulnerability in the PJSIP channel driver in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1, when using the res_pjsip_refer module, allows remote attackers to cause a denial of service (crash) via an in-dialog INVITE with Replaces message, which triggers the channel to be hung up.

Race condition in the chan_pjsip channel driver in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1 allows remote attackers to cause a denial of service (assertion failure and crash) via a cancel request for a SIP session with a queued action to (1) answer a session or (2) send ringing.

Incorrect access controls in res_srtp.c in Sangoma Asterisk 13.38.1, 16.16.0, 17.9.1, and 18.2.0 and Certified Asterisk 16.8-cert5 allow a remote unauthenticated attacker to prematurely terminate secure calls by replaying SRTP packets.

An issue was discovered in Sangoma Asterisk 16.x before 16.16.1, 17.x before 17.9.2, and 18.x before 18.2.1 and Certified Asterisk before 16.8-cert6. When re-negotiating for T.38, if the initial remote response was delayed just enough, Asterisk would send both audio and T.38 in the SDP. If this happened, and the remote responded with a declined T.38 stream, then Asterisk would crash.

The SIP channel driver (channels/chan_sip.c) in Asterisk Open Source 1.8.x before 1.8.23.1, 10.x before 10.12.3, and 11.x before 11.5.1; Certified Asterisk 1.8.15 before 1.8.15-cert3 and 11.2 before 11.2-cert2; and Asterisk Digiumphones 10.x-digiumphones before 10.12.3-digiumphones allows remote attackers to cause a denial of service (NULL pointer dereference, segmentation fault, and daemon crash) via an invalid SDP that defines a media description before the connection description in a SIP request.

The multi-part body parser in PJSIP, as used in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1, Certified Asterisk 13.13 before 13.13-cert4, and other products, allows remote attackers to cause a denial of service (out-of-bounds read and application crash) via a crafted packet.

The SIP channel driver (channels/chan_sip.c) in Asterisk Open Source 1.8.17.x through 1.8.22.x, 1.8.23.x before 1.8.23.1, and 11.x before 11.5.1 and Certified Asterisk 1.8.15 before 1.8.15-cert3 and 11.2 before 11.2-cert2 allows remote attackers to cause a denial of service (NULL pointer dereference, segmentation fault, and daemon crash) via an ACK with SDP to a previously terminated channel. NOTE: some of these details are obtained from third party information.

Multiple stack consumption vulnerabilities in Asterisk Open Source 1.8.x before 1.8.19.1, 10.x before 10.11.1, and 11.x before 11.1.2; Certified Asterisk 1.8.11 before 1.8.11-cert10; and Asterisk Digiumphones 10.x-digiumphones before 10.11.1-digiumphones allow remote attackers to cause a denial of service (daemon crash) via TCP data using the (1) SIP, (2) HTTP, or (3) XMPP protocol.

Asterisk Open Source 1.8.x before 1.8.28.1, 11.x before 11.10.1, and 12.x before 12.3.1 and Certified Asterisk 1.8.15 before 1.8.15-cert6 and 11.6 before 11.6-cert3 allows remote attackers to cause a denial of service (connection consumption) via a large number of (1) inactive or (2) incomplete HTTP connections.

An issue was discovered in Asterisk Open Source 13.12.x and 13.13.x before 13.13.1 and 14.x before 14.2.1. If an SDP offer or answer is received with the Opus codec and with the format parameters separated using a space the code responsible for parsing will recursively call itself until it crashes. This occurs as the code does not properly handle spaces separating the parameters. This does NOT require the endpoint to have Opus configured in Asterisk. This also does not require the endpoint to be authenticated. If guest is enabled for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still processed and the crash occurs.

An issue was discovered in Asterisk through 19.x. When using STIR/SHAKEN, it is possible to download files that are not certificates. These files could be much larger than what one would expect to download, leading to Resource Exhaustion. This is fixed in 16.25.2, 18.11.2, and 19.3.2.

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.

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.

chan_sip.c in the SIP channel driver in Asterisk Open Source 1.6.x before 1.6.2.18.1 and 1.8.x before 1.8.4.3 does not properly handle '\0' characters in SIP packets, which allows remote attackers to cause a denial of service (memory corruption) or possibly have unspecified other impact via a crafted packet.

reqresp_parser.c in the SIP channel driver in Asterisk Open Source 1.8.x before 1.8.4.2 does not initialize certain strings, which allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a malformed Contact header.

In the pjsip channel driver (res_pjsip) in Asterisk 13.x before 13.17.1 and 14.x before 14.6.1, a carefully crafted tel URI in a From, To, or Contact header could cause Asterisk to crash.

Asterisk Open Source 1.4.x before 1.4.40.1, 1.6.1.x before 1.6.1.25, 1.6.2.x before 1.6.2.17.3, and 1.8.x before 1.8.3.3 and Asterisk Business Edition C.x.x before C.3.6.4 do not restrict the number of unauthenticated sessions to certain interfaces, which allows remote attackers to cause a denial of service (file descriptor exhaustion and disk space exhaustion) via a series of TCP connections.

chan_iax2.c in the IAX2 channel driver in Asterisk Open Source 1.4.x before 1.4.41.1, 1.6.2.x before 1.6.2.18.1, and 1.8.x before 1.8.4.3, and Asterisk Business Edition C.3 before C.3.7.3, accesses a memory address contained in an option control frame, which allows remote attackers to cause a denial of service (daemon crash) or possibly have unspecified other impact via a crafted frame.

chain_sip in Asterisk Open Source 11.x before 11.23.1 and 13.x 13.11.1 and Certified Asterisk 11.6 before 11.6-cert15 and 13.8 before 13.8-cert3 allows remote attackers to cause a denial of service (port exhaustion).

main/translate.c in Sangoma Asterisk 13.28.0 and 16.5.0 allows a remote attacker to send a specific RTP packet during a call and cause a crash in a specific scenario.

Buffer overflow in the unpacksms16 function in apps/app_sms.c in Asterisk Open Source 1.8.x before 1.8.24.1, 10.x before 10.12.4, and 11.x before 11.6.1; Asterisk with Digiumphones 10.x-digiumphones before 10.12.4-digiumphones; and Certified Asterisk 1.8.x before 1.8.15-cert4 and 11.x before 11.2-cert3 allows remote attackers to cause a denial of service (daemon crash) via a 16-bit SMS message with an odd number of bytes, which triggers an infinite loop.

PJSIP, as used in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1, Certified Asterisk 13.13 before 13.13-cert4, and other products, allows remote attackers to cause a denial of service (buffer overflow and application crash) via a SIP packet with a crafted CSeq header in conjunction with a Via header that lacks a branch parameter.

res_pjsip_t38 in Sangoma Asterisk 16.x before 16.16.2, 17.x before 17.9.3, and 18.x before 18.2.2, and Certified Asterisk before 16.8-cert7, allows an attacker to trigger a crash by sending an m=image line and zero port in a response to a T.38 re-invite initiated by Asterisk. This is a re-occurrence of the CVE-2019-15297 symptoms but not for exactly the same reason. The crash occurs because there is an append operation relative to the active topology, but this should instead be a replace operation.

res_pjsip_t38 in Sangoma Asterisk 15.x before 15.7.4 and 16.x before 16.5.1 allows an attacker to trigger a crash by sending a declined stream in a response to a T.38 re-invite initiated by Asterisk. The crash occurs because of a NULL session media object dereference.

An issue was discovered in Asterisk Open Source through 13.27.0, 14.x and 15.x through 15.7.2, and 16.x through 16.4.0, and Certified Asterisk through 13.21-cert3. A pointer dereference in chan_sip while handling SDP negotiation allows an attacker to crash Asterisk when handling an SDP answer to an outgoing T.38 re-invite. To exploit this vulnerability an attacker must cause the chan_sip module to send a T.38 re-invite request to them. Upon receipt, the attacker must send an SDP answer containing both a T.38 UDPTL stream and another media stream containing only a codec (which is not permitted according to the chan_sip configuration).

The _xrealloc function in xlsp_xmalloc.c in OpenSLP 2.0.0 allows remote attackers to cause a denial of service (NULL pointer dereference and crash) via a large number of crafted packets, which triggers a memory allocation failure.

In Apache Subversion versions up to and including 1.9.10, 1.10.4, 1.12.0, Subversion's svnserve server process may exit when a client sends certain sequences of protocol commands. This can lead to disruption for users of the server.

SAP CommonCryptoLib version 8.5.38 or lower is vulnerable to null pointer dereference vulnerability when an unauthenticated attacker sends crafted malicious data in the HTTP requests over the network, this causes the SAP application to crash and has high impact on the availability of the SAP system.

The mod_dav_svn module for the Apache HTTP Server, as distributed in Apache Subversion before 1.6.17, allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a request for a baselined WebDAV resource, as exploited in the wild in May 2011.

ncmpc through 0.29 is prone to a NULL pointer dereference flaw. If a user uses the chat screen and another client sends a long chat message, a crash and denial of service could occur.

The ip_evictor function in ip_fragment.c in libnids before 1.24, as used in dsniff and possibly other products, allows remote attackers to cause a denial of service (NULL pointer dereference and crash) via crafted fragmented packets.

IP fragmentation denial of service in FreeBSD allows a remote attacker to cause a crash.

An issue was discovered in Irssi before 1.0.7 and 1.1.x before 1.1.1. When the number of windows exceeds the available space, a crash due to a NULL pointer dereference would occur.

In Ceph before 12.2.3 and 13.x through 13.0.1, the rgw_civetweb.cc RGWCivetWeb::init_env function in radosgw doesn't handle malformed HTTP headers properly, allowing for denial of service.

A NULL pointer dereference flaw was found in the way LibVNCServer before 0.9.9 handled certain ClientCutText message. A remote attacker could use this flaw to crash the VNC server by sending a specially crafted ClientCutText message from a VNC client.

It was found in FreeBSD 8.0, 6.3 and 4.9, and OpenBSD 4.6 that a null pointer dereference in ftpd/popen.c may lead to remote denial of service of the ftpd service.

SBLIM Small Footprint CIM Broker (SFCB) 1.4.9 has a null pointer (DoS) vulnerability via a crafted POST request to the /cimom URI.