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.

A Buffer Overflow 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. No size checking is done when setting the user field for Party B on a CDR. Thus, it is possible for someone to use an arbitrarily large string and write past the end of the user field storage buffer. NOTE: this is different from CVE-2017-7617, which was only about the Party A buffer.

In Asterisk 11.x before 11.25.3, 13.x before 13.17.2, and 14.x before 14.6.2 and Certified Asterisk 11.x before 11.6-cert18 and 13.x before 13.13-cert6, insufficient RTCP packet validation could allow reading stale buffer contents and when combined with the "nat" and "symmetric_rtp" options allow redirecting where Asterisk sends the next RTCP report.

In Asterisk 11.x before 11.25.2, 13.x before 13.17.1, and 14.x before 14.6.1 and Certified Asterisk 11.x before 11.6-cert17 and 13.x before 13.13-cert5, unauthorized command execution is possible. The app_minivm module has an "externnotify" program configuration option that is executed by the MinivmNotify dialplan application. The application uses the caller-id name and number as part of a built string passed to the OS shell for interpretation and execution. Since the caller-id name and number can come from an untrusted source, a crafted caller-id name or number allows an arbitrary shell command injection.

In res/res_rtp_asterisk.c in Asterisk 11.x before 11.25.2, 13.x before 13.17.1, and 14.x before 14.6.1 and Certified Asterisk 11.x before 11.6-cert17 and 13.x before 13.13-cert5, unauthorized data disclosure (media takeover in the RTP stack) is possible with careful timing by an attacker. The "strictrtp" option in rtp.conf enables a feature of the RTP stack that learns the source address of media for a session and drops any packets that do not originate from the expected address. This option is enabled by default in Asterisk 11 and above. The "nat" and "rtp_symmetric" options (for chan_sip and chan_pjsip, respectively) enable symmetric RTP support in the RTP stack. This uses the source address of incoming media as the target address of any sent media. This option is not enabled by default, but is commonly enabled to handle devices behind NAT. A change was made to the strict RTP support in the RTP stack to better tolerate late media when a reinvite occurs. When combined with the symmetric RTP support, this introduced an avenue where media could be hijacked. Instead of only learning a new address when expected, the new code allowed a new source address to be learned at all times. If a flood of RTP traffic was received, the strict RTP support would allow the new address to provide media, and (with symmetric RTP enabled) outgoing traffic would be sent to this new address, allowing the media to be hijacked. Provided the attacker continued to send traffic, they would continue to receive traffic as well.

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.

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).

Remote code execution can occur in Asterisk Open Source 13.x before 13.14.1 and 14.x before 14.3.1 and Certified Asterisk 13.13 before 13.13-cert3 because of a buffer overflow in a CDR user field, related to X-ClientCode in chan_sip, the CDR dialplan function, and the AMI Monitor action.

An issue was discovered in Asterisk Open Source 11.x before 11.25.1, 13.x before 13.13.1, and 14.x before 14.2.1 and Certified Asterisk 11.x before 11.6-cert16 and 13.x before 13.8-cert4. The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue does not affect you.

chan_sip in Asterisk Open Source 1.8.x, 11.x before 11.21.1, 12.x, and 13.x before 13.7.1 and Certified Asterisk 1.8.28, 11.6 before 11.6-cert12, and 13.1 before 13.1-cert3, when the timert1 sip.conf configuration is set to a value greater than 1245, allows remote attackers to cause a denial of service (file descriptor consumption) via vectors related to large retransmit timeout values.

Asterisk Open Source 1.8.x, 11.x before 11.21.1, 12.x, and 13.x before 13.7.1 and Certified Asterisk 1.8.28, 11.6 before 11.6-cert12, and 13.1 before 13.1-cert3 allow remote authenticated users to cause a denial of service (uninitialized pointer dereference and crash) via a zero length error correcting redundancy packet for a UDPTL FAX packet that is lost.

Asterisk Open Source 1.8 before 1.8.32.3, 11.x before 11.17.1, 12.x before 12.8.2, and 13.x before 13.3.2 and Certified Asterisk 1.8.28 before 1.8.28-cert5, 11.6 before 11.6-cert11, and 13.1 before 13.1-cert2, when registering a SIP TLS device, does not properly handle a null byte in a domain name in the subject's Common Name (CN) field of an X.509 certificate, which allows man-in-the-middle attackers to spoof arbitrary SSL servers via a crafted certificate issued by a legitimate Certification Authority.

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.

Stack-based buffer overflow in the ast_uri_encode function in main/utils.c in Asterisk Open Source before 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.1, 1.8.1.2, 1.8.2.; and Business Edition before C.3.6.2; when running in pedantic mode allows remote authenticated users to execute arbitrary code via crafted caller ID data in vectors involving the (1) SIP channel driver, (2) URIENCODE dialplan function, or (3) AGI dialplan function.

The SIP channel driver in Asterisk Open Source 1.2.x before 1.2.34, 1.4.x before 1.4.26.1, 1.6.0.x before 1.6.0.12, and 1.6.1.x before 1.6.1.4; Asterisk Business Edition A.x.x, B.x.x before B.2.5.9, C.2.x before C.2.4.1, and C.3.x before C.3.1; and Asterisk Appliance s800i 1.2.x before 1.3.0.3 does not use a maximum width when invoking sscanf style functions, which allows remote attackers to cause a denial of service (stack memory consumption) via SIP packets containing large sequences of ASCII decimal characters, as demonstrated via vectors related to (1) the CSeq value in a SIP header, (2) large Content-Length value, and (3) SDP.