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.
References
| Link | Resource |
|---|---|
| http://downloads.asterisk.org/pub/security/AST-2016-009.html | Mitigation Vendor Advisory |
| http://www.securityfocus.com/bid/94789 | Third Party Advisory VDB Entry |
| http://www.securitytracker.com/id/1037408 |
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Configurations
Configuration 1 (hide)
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Configuration 2 (hide)
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Information
Published : 2016-12-12 13:59
Updated : 2017-07-26 18:29
NVD link : CVE-2016-9938
Mitre link : CVE-2016-9938
JSON object : View
CWE
CWE-285
Improper Authorization
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Products Affected
digium
- certified_asterisk
- asterisk