Digium Certified Asterisk 11.6 cert11 LTS

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.

Published: 2016-12-12T21:00:00.000Z
Updated: 2024-08-06T03:07:31.471Z

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).

Published: 2017-04-17T16:00:00.000Z
Updated: 2024-08-06T02:04:55.787Z

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.

Published: 2016-02-22T15:05:00.000Z
Updated: 2024-08-05T23:24:48.520Z

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.

Published: 2016-02-22T15:05:00.000Z
Updated: 2024-08-05T23:24:48.950Z