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.
chan_iax2.c in the IAX2 channel driver in Certified Asterisk 1.8.11-cert before 1.8.11-cert2 and Asterisk Open Source 1.8.x before 1.8.12.1 and 10.x before 10.4.1, when a certain mohinterpret setting is enabled, allows remote attackers to cause a denial of service (daemon crash) by placing a call on hold.
The SIP over UDP implementation in Asterisk Open Source 1.4.x before 1.4.43, 1.6.x before 1.6.2.21, and 1.8.x before 1.8.7.2 uses different port numbers for responses to invalid requests depending on whether a SIP username exists, which allows remote attackers to enumerate usernames via a series of requests.
The handle_request_info function in channels/chan_sip.c in Asterisk Open Source 1.6.2.x before 1.6.2.21 and 1.8.x before 1.8.7.2, when automon is enabled, allows remote attackers to cause a denial of service (NULL pointer dereference and daemon crash) via a crafted sequence of SIP requests.