A res_pjsip_session crash exists in Asterisk Open Source 13.x prior to 13.37.1, 16.x prior to 16.14.1, 17.x prior to 17.8.1, and 18.x prior to 18.0.1. and Certified Asterisk prior to 16.8-cert5. Upon receiving a new SIP Invite, Asterisk did not return the created dialog locked or referenced. This caused a gap between the creation of the dialog object, and its next use by the thread that created it. Depending on some off-nominal circumstances and timing, it was possible for another thread to free said dialog in this gap. Asterisk could then crash when the dialog object, or any of its dependent objects, were dereferenced or accessed next by the initial-creation thread. Note, however, that this crash can only occur when using a connection-oriented protocol (e.g., TCP or TLS, but not UDP) for SIP transport. Also, the remote client must be authenticated, or Asterisk must be configured for anonymous calling.
| Vulnerable Product | Search on Vulmon | Subscribe to Product |
|---|---|---|
asterisk open source |
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digium certified asterisk 16.8 |
Vulmon