channels/chan_sip.c in Asterisk Open Source 1.8.x before 1.8.13.1 and 10.x before 10.5.2, Asterisk Business Edition C.3.x before C.3.7.5, Certified Asterisk 1.8.11-certx before 1.8.11-cert4, and Asterisk Digiumphones 10.x.x-digiumphones before 10.5.2-digiumphones does not properly handle a provisional response to a SIP reINVITE request, which allows remote authenticated users to cause a denial of service (RTP port exhaustion) via sessions that lack final responses.
| Name | Vendor | Start Version | End Version |
|---|---|---|---|
| Asterisk_business_edition | Digium | c.3.1 (including) | c.3.1 (including) |
| Asterisk_business_edition | Digium | c.3.3 (including) | c.3.3 (including) |
| Asterisk_business_edition | Digium | c.3.7.4 (including) | c.3.7.4 (including) |
| Asterisk | Ubuntu | hardy | * |
| Asterisk | Ubuntu | lucid | * |
| Asterisk | Ubuntu | natty | * |
| Asterisk | Ubuntu | oneiric | * |
| Asterisk | Ubuntu | precise | * |
| Asterisk | Ubuntu | upstream | * |