Potential infinite loop when parsing WAV format file in PJSIP
PJSIP is a free and open source multimedia communication library written in C. A denial-of-service vulnerability affects applications on a 32-bit systems that use PJSIP versions 2.12 and prior to play/read invalid WAV files. The vulnerability occurs when reading WAV file data chunks with length greater than 31-bit integers. The vulnerability does not affect 64-bit apps and should not affect apps that only plays trusted WAV files. A patch is available on the `master` branch of the `pjsip/project` GitHub repository. As a workaround, apps can reject a WAV file received from an unknown source or validate the file first.
An issue was discovered in Asterisk through 19.x and Certified Asterisk through 16.8-cert13. The func_odbc module provides possibly inadequate escaping functionality for backslash characters in SQL queries, resulting in user-provided data creating a broken SQL query or possibly a SQL injection. This is fixed in 16.25.2, 18.11.2, and 19.3.2, and 16.8-cert14.
An SSRF issue was discovered in Asterisk through 19.x. When using STIR/SHAKEN, it's possible to send arbitrary requests (such as GET) to interfaces such as localhost by using the Identity header. This is fixed in 16.25.2, 18.11.2, and 19.3.2.
An issue was discovered in Asterisk through 19.x. When using STIR/SHAKEN, it is possible to download files that are not certificates. These files could be much larger than what one would expect to download, leading to Resource Exhaustion. This is fixed in 16.25.2, 18.11.2, and 19.3.2.
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In versions up to and including 2.11.1 when in a dialog set (or forking) scenario, a hash key shared by multiple UAC dialogs can potentially be prematurely freed when one of the dialogs is destroyed . The issue may cause a dialog set to be registered in the hash table multiple times (with different hash keys) leading to undefined behavior such as dialog list collision which eventually leading to endless loop. A patch is available in commit db3235953baa56d2fb0e276ca510fefca7516
Buffer overflow in PJSUA API when calling pjsua_call_dump. An attacker-controlled 'buffer' argument may cause a buffer overflow, since supplying an output buffer smaller than 128 characters may overflow the output buffer, regardless of the 'maxlen' argument supplied
Read out-of-bounds in PJSUA API when calling pjsua_recorder_create. An attacker-controlled 'filename' argument may cause an out-of-bounds read when the filename is shorter than 4 characters.
Stack overflow in PJSUA API when calling pjsua_playlist_create. An attacker-controlled 'file_names' argument may cause a buffer overflow since it is copied to a fixed-size stack buffer without any size validation.
Stack overflow in PJSUA API when calling pjsua_recorder_create. An attacker-controlled 'filename' argument may cause a buffer overflow since it is copied to a fixed-size stack buffer without any size validation.
Stack overflow in PJSUA API when calling pjsua_player_create. An attacker-controlled 'filename' argument may cause a buffer overflow since it is copied to a fixed-size stack buffer without any size validation.
PJSIP is a free and open source multimedia communication library. In version 2.11.1 and prior, if incoming RTCP XR message contain block, the data field is not checked against the received packet size, potentially resulting in an out-of-bound read access. This affects all users that use PJMEDIA and RTCP XR. A malicious actor can send a RTCP XR message with an invalid packet size.
An issue was discovered in Sangoma Asterisk 13.x before 13.38.3, 16.x before 16.19.1, 17.x before 17.9.4, and 18.x before 18.5.1, and Certified Asterisk before 16.8-cert10. If the IAX2 channel driver receives a packet that contains an unsupported media format, a crash can occur.
An issue was discovered in PJSIP in Asterisk before 16.19.1 and before 18.5.1. To exploit, a re-INVITE without SDP must be received after Asterisk has sent a BYE request.
Incorrect access controls in res_srtp.c in Sangoma Asterisk 13.38.1, 16.16.0, 17.9.1, and 18.2.0 and Certified Asterisk 16.8-cert5 allow a remote unauthenticated attacker to prematurely terminate secure calls by replaying SRTP packets.
A buffer overflow in res_pjsip_diversion.c in Sangoma Asterisk versions 13.38.1, 16.15.1, 17.9.1, and 18.1.1 allows remote attacker to crash Asterisk by deliberately misusing SIP 181 responses.
An issue was discovered in res_pjsip_diversion.c in Sangoma Asterisk before 13.38.0, 14.x through 16.x before 16.15.0, 17.x before 17.9.0, and 18.x before 18.1.0. A crash can occur when a SIP message is received with a History-Info header that contains a tel-uri, or when a SIP 181 response is received that contains a tel-uri in the Diversion header.
A res_pjsip_session crash was discovered in Asterisk Open Source 13.x before 13.37.1, 16.x before 16.14.1, 17.x before 17.8.1, and 18.x before 18.0.1. and Certified Asterisk before 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 i
An issue was discovered in Asterisk Open Source 13.x before 13.37.1, 16.x before 16.14.1, 17.x before 17.8.1, and 18.x before 18.0.1 and Certified Asterisk before 16.8-cert5. If Asterisk is challenged on an outbound INVITE and the nonce is changed in each response, Asterisk will continually send INVITEs in a loop. This causes Asterisk to consume more and more memory since the transaction will never terminate (even if the call is hung up), ultimately leading to a restart or shutdown of Asterisk. Outbound authentication must be configured on the endpoint for this to occur.
An issue was discovered in manager.c in Sangoma Asterisk through 13.x, 16.x, 17.x and Certified Asterisk 13.21 through 13.21-cert4. A remote authenticated Asterisk Manager Interface (AMI) user without system authorization could use a specially crafted Originate AMI request to execute arbitrary system commands.
An issue was discovered in channels/chan_sip.c in Sangoma Asterisk 13.x before 13.29.2, 16.x before 16.6.2, and 17.x before 17.0.1, and Certified Asterisk 13.21 before cert5. A SIP request can be sent to Asterisk that can change a SIP peer's IP address. A REGISTER does not need to occur, and calls can be hijacked as a result. The only thing that needs to be known is the peer's name; authentication details such as passwords do not need to be known. This vulnerability is only exploitable when the nat option is set to the default, or auto_force_rport.