Search Results
There are 65 CVE Records that match your search.
| Name | Description |
|---|---|
| CVE-2023-38703 | PJSIP is a free and open source multimedia communication library written in C with high level API in C, C++, Java, C#, and Python languages. SRTP is a higher level media transport which is stacked upon a lower level media transport such as UDP and ICE. Currently a higher level transport is not synchronized with its lower level transport that may introduce use-after-free issue. This vulnerability affects applications that have SRTP capability (`PJMEDIA_HAS_SRTP` is set) and use underlying media transport other than UDP. This vulnerability’s impact may range from unexpected application termination to control flow hijack/memory corruption. The patch is available as a commit in the master branch. |
| CVE-2023-37457 | Asterisk is an open source private branch exchange and telephony toolkit. In Asterisk versions 18.20.0 and prior, 20.5.0 and prior, and 21.0.0; as well as ceritifed-asterisk 18.9-cert5 and prior, the 'update' functionality of the PJSIP_HEADER dialplan function can exceed the available buffer space for storing the new value of a header. By doing so this can overwrite memory or cause a crash. This is not externally exploitable, unless dialplan is explicitly written to update a header based on data from an outside source. If the 'update' functionality is not used the vulnerability does not occur. A patch is available at commit a1ca0268254374b515fa5992f01340f7717113fa. |
| CVE-2023-27585 | PJSIP is a free and open source multimedia communication library written in C. A buffer overflow vulnerability in versions 2.13 and prior affects applications that use PJSIP DNS resolver. It doesn't affect PJSIP users who do not utilise PJSIP DNS resolver. This vulnerability is related to CVE-2022-24793. The difference is that this issue is in parsing the query record `parse_query()`, while the issue in CVE-2022-24793 is in `parse_rr()`. A patch is available as commit `d1c5e4d` in the `master` branch. A workaround is to disable DNS resolution in PJSIP config (by setting `nameserver_count` to zero) or use an external resolver implementation instead. |
| CVE-2022-42705 | A use-after-free in res_pjsip_pubsub.c in Sangoma Asterisk 16.28, 18.14, 19.6, and certified/18.9-cert2 may allow a remote authenticated attacker to crash Asterisk (denial of service) by performing activity on a subscription via a reliable transport at the same time that Asterisk is also performing activity on that subscription. |
| CVE-2022-39269 | PJSIP is a free and open source multimedia communication library written in C. When processing certain packets, PJSIP may incorrectly switch from using SRTP media transport to using basic RTP upon SRTP restart, causing the media to be sent insecurely. The vulnerability impacts all PJSIP users that use SRTP. The patch is available as commit d2acb9a in the master branch of the project and will be included in version 2.13. Users are advised to manually patch or to upgrade. There are no known workarounds for this vulnerability. |
| CVE-2022-39244 | PJSIP is a free and open source multimedia communication library written in C. In versions of PJSIP prior to 2.13 the PJSIP parser, PJMEDIA RTP decoder, and PJMEDIA SDP parser are affeced by a buffer overflow vulnerability. Users connecting to untrusted clients are at risk. This issue has been patched and is available as commit c4d3498 in the master branch and will be included in releases 2.13 and later. Users are advised to upgrade. There are no known workarounds for this issue. |
| CVE-2022-31031 | 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 prior to and including 2.12.1 a stack buffer overflow vulnerability affects PJSIP users that use STUN in their applications, either by: setting a STUN server in their account/media config in PJSUA/PJSUA2 level, or directly using `pjlib-util/stun_simple` API. A patch is available in commit 450baca which should be included in the next release. There are no known workarounds for this issue. |
| CVE-2022-29330 | Missing access control in the backup system of Telesoft VitalPBX before 3.2.1 allows attackers to access the PJSIP and SIP extension credentials, cryptographic keys and voicemails files via unspecified vectors. |
| CVE-2022-24793 | PJSIP is a free and open source multimedia communication library written in C. A buffer overflow vulnerability in versions 2.12 and prior affects applications that use PJSIP DNS resolution. It doesn't affect PJSIP users who utilize an external resolver. This vulnerability is related to CVE-2023-27585. The difference is that this issue is in parsing the query record `parse_rr()`, while the issue in CVE-2023-27585 is in `parse_query()`. A patch is available in the `master` branch of the `pjsip/pjproject` GitHub repository. A workaround is to disable DNS resolution in PJSIP config (by setting `nameserver_count` to zero) or use an external resolver instead. |
| CVE-2022-24792 | 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. |
| CVE-2022-24786 | PJSIP is a free and open source multimedia communication library written in C. PJSIP versions 2.12 and prior do not parse incoming RTCP feedback RPSI (Reference Picture Selection Indication) packet, but any app that directly uses pjmedia_rtcp_fb_parse_rpsi() will be affected. A patch is available in the `master` branch of the `pjsip/pjproject` GitHub repository. There are currently no known workarounds. |
| CVE-2022-24764 | PJSIP is a free and open source multimedia communication library written in C. Versions 2.12 and prior contain a stack buffer overflow vulnerability that affects PJSUA2 users or users that call the API `pjmedia_sdp_print(), pjmedia_sdp_media_print()`. Applications that do not use PJSUA2 and do not directly call `pjmedia_sdp_print()` or `pjmedia_sdp_media_print()` should not be affected. A patch is available on the `master` branch of the `pjsip/pjproject` GitHub repository. There are currently no known workarounds. |
| CVE-2022-24763 | PJSIP is a free and open source multimedia communication library written in the C language. Versions 2.12 and prior contain a denial-of-service vulnerability that affects PJSIP users that consume PJSIP's XML parsing in their apps. Users are advised to update. There are no known workarounds. |
| CVE-2022-24754 | PJSIP is a free and open source multimedia communication library written in C language. In versions prior to and including 2.12 PJSIP there is a stack-buffer overflow vulnerability which only impacts PJSIP users who accept hashed digest credentials (credentials with data_type `PJSIP_CRED_DATA_DIGEST`). This issue has been patched in the master branch of the PJSIP repository and will be included with the next release. Users unable to upgrade need to check that the hashed digest data length must be equal to `PJSIP_MD5STRLEN` before passing to PJSIP. |
| CVE-2022-23608 | 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 db3235953baa56d2fb0e276ca510fefca751643f which will be included in the next release. There are no known workarounds for this issue. |
| CVE-2022-23547 | 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. This issue is similar to GHSA-9pfh-r8x4-w26w. Possible buffer overread when parsing a certain STUN message. The vulnerability affects applications that uses STUN including PJNATH and PJSUA-LIB. The patch is available as commit in the master branch. |
| CVE-2022-23537 | 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. Buffer overread is possible when parsing a specially crafted STUN message with unknown attribute. The vulnerability affects applications that uses STUN including PJNATH and PJSUA-LIB. The patch is available as a commit in the master branch (2.13.1). |
| CVE-2022-21723 | 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 2.11.1 and prior, parsing an incoming SIP message that contains a malformed multipart can potentially cause out-of-bound read access. This issue affects all PJSIP users that accept SIP multipart. The patch is available as commit in the `master` branch. There are no known workarounds. |
| CVE-2022-21722 | 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 version 2.11.1 and prior, there are various cases where it is possible that certain incoming RTP/RTCP packets can potentially cause out-of-bound read access. This issue affects all users that use PJMEDIA and accept incoming RTP/RTCP. A patch is available as a commit in the `master` branch. There are no known workarounds. |
| CVE-2021-46837 | res_pjsip_t38 in Sangoma Asterisk 16.x before 16.16.2, 17.x before 17.9.3, and 18.x before 18.2.2, and Certified Asterisk before 16.8-cert7, allows an attacker to trigger a crash by sending an m=image line and zero port in a response to a T.38 re-invite initiated by Asterisk. This is a re-occurrence of the CVE-2019-15297 symptoms but not for exactly the same reason. The crash occurs because there is an append operation relative to the active topology, but this should instead be a replace operation. |
| CVE-2021-43845 | 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. |
| CVE-2021-43804 | 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 affected versions if the incoming RTCP BYE message contains a reason's length, this declared length is not checked against the actual received packet size, potentially resulting in an out-of-bound read access. This issue affects all users that use PJMEDIA and RTCP. A malicious actor can send a RTCP BYE message with an invalid reason length. Users are advised to upgrade as soon as possible. There are no known workarounds. |
| CVE-2021-43303 | 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 |
| CVE-2021-43302 | 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. |
| CVE-2021-43301 | 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. |
| CVE-2021-43300 | 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. |
| CVE-2021-43299 | 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. |
| CVE-2021-41141 | PJSIP is a free and open source multimedia communication library written in the C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. In various parts of PJSIP, when error/failure occurs, it is found that the function returns without releasing the currently held locks. This could result in a system deadlock, which cause a denial of service for the users. No release has yet been made which contains the linked fix commit. All versions up to an including 2.11.1 are affected. Users may need to manually apply the patch. |
| CVE-2021-37706 | 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 affected versions if the incoming STUN message contains an ERROR-CODE attribute, the header length is not checked before performing a subtraction operation, potentially resulting in an integer underflow scenario. This issue affects all users that use STUN. A malicious actor located within the victim’s network may forge and send a specially crafted UDP (STUN) message that could remotely execute arbitrary code on the victim’s machine. Users are advised to upgrade as soon as possible. There are no known workarounds. |
| CVE-2021-32686 | 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 PJSIP before version 2.11.1, there are a couple of issues found in the SSL socket. First, a race condition between callback and destroy, due to the accepted socket having no group lock. Second, the SSL socket parent/listener may get destroyed during handshake. Both issues were reported to happen intermittently in heavy load TLS connections. They cause a crash, resulting in a denial of service. These are fixed in version 2.11.1. |
| CVE-2021-31878 | 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. |
| CVE-2021-26906 | An issue was discovered in res_pjsip_session.c in Digium Asterisk through 13.38.1; 14.x, 15.x, and 16.x through 16.16.0; 17.x through 17.9.1; and 18.x through 18.2.0, and Certified Asterisk through 16.8-cert5. An SDP negotiation vulnerability in PJSIP allows a remote server to potentially crash Asterisk by sending specific SIP responses that cause an SDP negotiation failure. |
| CVE-2021-21375 | 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 PJSIP version 2.10 and earlier, after an initial INVITE has been sent, when two 183 responses are received, with the first one causing negotiation failure, a crash will occur. This results in a denial of service. |
| CVE-2020-35776 | 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. |
| CVE-2020-35652 | 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. |
| CVE-2020-28327 | 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 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. |
| CVE-2020-15260 | 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 version 2.10 and earlier, PJSIP transport can be reused if they have the same IP address + port + protocol. However, this is insufficient for secure transport since it lacks remote hostname authentication. Suppose we have created a TLS connection to `sip.foo.com`, which has an IP address `100.1.1.1`. If we want to create a TLS connection to another hostname, say `sip.bar.com`, which has the same IP address, then it will reuse that existing connection, even though `100.1.1.1` does not have certificate to authenticate as `sip.bar.com`. The vulnerability allows for an insecure interaction without user awareness. It affects users who need access to connections to different destinations that translate to the same address, and allows man-in-the-middle attack if attacker can route a connection to another destination such as in the case of DNS spoofing. |
| CVE-2019-7251 | An Integer Signedness issue (for a return code) in the res_pjsip_sdp_rtp module in Digium Asterisk versions 15.7.1 and earlier and 16.1.1 and earlier allows remote authenticated users to crash Asterisk via a specially crafted SDP protocol violation. |
| CVE-2019-18976 | An issue was discovered in res_pjsip_t38.c in Sangoma Asterisk through 13.x and Certified Asterisk through 13.21-x. If it receives a re-invite initiating T.38 faxing and has a port of 0 and no c line in the SDP, a NULL pointer dereference and crash will occur. This is different from CVE-2019-18940. |
| CVE-2019-15297 | res_pjsip_t38 in Sangoma Asterisk 15.x before 15.7.4 and 16.x before 16.5.1 allows an attacker to trigger a crash by sending a declined stream in a response to a T.38 re-invite initiated by Asterisk. The crash occurs because of a NULL session media object dereference. |
| CVE-2019-12827 | Buffer overflow in res_pjsip_messaging in Digium Asterisk versions 13.21-cert3, 13.27.0, 15.7.2, 16.4.0 and earlier allows remote authenticated users to crash Asterisk by sending a specially crafted SIP MESSAGE message. |
| CVE-2018-7286 | An issue was discovered in Asterisk through 13.19.1, 14.x through 14.7.5, and 15.x through 15.2.1, and Certified Asterisk through 13.18-cert2. res_pjsip allows remote authenticated users to crash Asterisk (segmentation fault) by sending a number of SIP INVITE messages on a TCP or TLS connection and then suddenly closing the connection. |
| CVE-2018-7284 | A Buffer Overflow issue was discovered in Asterisk through 13.19.1, 14.x through 14.7.5, and 15.x through 15.2.1, and Certified Asterisk through 13.18-cert2. When processing a SUBSCRIBE request, the res_pjsip_pubsub module stores the accepted formats present in the Accept headers of the request. This code did not limit the number of headers it processed, despite having a fixed limit of 32. If more than 32 Accept headers were present, the code would write outside of its memory and cause a crash. |
| CVE-2018-1000099 | Teluu PJSIP version 2.7.1 and earlier contains a Access of Null/Uninitialized Pointer vulnerability in pjmedia SDP parsing that can result in Crash. This attack appear to be exploitable via Sending a specially crafted message. This vulnerability appears to have been fixed in 2.7.2. |
| CVE-2018-1000098 | Teluu PJSIP version 2.7.1 and earlier contains a Integer Overflow vulnerability in pjmedia SDP parsing that can result in Crash. This attack appear to be exploitable via Sending a specially crafted message. This vulnerability appears to have been fixed in 2.7.2. |
| CVE-2017-9372 | PJSIP, as used in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1, Certified Asterisk 13.13 before 13.13-cert4, and other products, allows remote attackers to cause a denial of service (buffer overflow and application crash) via a SIP packet with a crafted CSeq header in conjunction with a Via header that lacks a branch parameter. |
| CVE-2017-9359 | The multi-part body parser in PJSIP, as used in Asterisk Open Source 13.x before 13.15.1 and 14.x before 14.4.1, Certified Asterisk 13.13 before 13.13-cert4, and other products, allows remote attackers to cause a denial of service (out-of-bounds read and application crash) via a crafted packet. |
| CVE-2017-17850 | An issue was discovered in Asterisk 13.18.4 and older, 14.7.4 and older, 15.1.4 and older, and 13.18-cert1 and older. A select set of SIP messages create a dialog in Asterisk. Those SIP messages must contain a contact header. For those messages, if the header was not present and the PJSIP channel driver was used, Asterisk would crash. The severity of this vulnerability is somewhat mitigated if authentication is enabled. If authentication is enabled, a user would have to first be authorized before reaching the crash point. |
| CVE-2017-16875 | An issue was discovered in Teluu pjproject (pjlib and pjlib-util) in PJSIP before 2.7.1. The ioqueue component may issue a double key unregistration after an attacker initiates a socket connection with specific settings and sequences. Such double key unregistration will trigger an integer overflow, which may cause ioqueue backends to reject future key registrations. |
| CVE-2017-16872 | An issue was discovered in Teluu pjproject (pjlib and pjlib-util) in PJSIP before 2.7.1. Parsing the numeric header fields in a SIP message (like cseq, ttl, port, etc.) all had the potential to overflow, either causing unintended values to be captured or, if the values were subsequently converted back to strings, a buffer overrun. This will lead to a potential exploit using carefully crafted invalid values. |
| CVE-2017-16672 | An issue was discovered in Asterisk Open Source 13 before 13.18.1, 14 before 14.7.1, and 15 before 15.1.1 and Certified Asterisk 13.13 before 13.13-cert7. A memory leak occurs when an Asterisk pjsip session object is created and that call gets rejected before the session itself is fully established. When this happens the session object never gets destroyed. Eventually Asterisk can run out of memory and crash. |
| CVE-2017-14099 | In res/res_rtp_asterisk.c in Asterisk 11.x before 11.25.2, 13.x before 13.17.1, and 14.x before 14.6.1 and Certified Asterisk 11.x before 11.6-cert17 and 13.x before 13.13-cert5, unauthorized data disclosure (media takeover in the RTP stack) is possible with careful timing by an attacker. The "strictrtp" option in rtp.conf enables a feature of the RTP stack that learns the source address of media for a session and drops any packets that do not originate from the expected address. This option is enabled by default in Asterisk 11 and above. The "nat" and "rtp_symmetric" options (for chan_sip and chan_pjsip, respectively) enable symmetric RTP support in the RTP stack. This uses the source address of incoming media as the target address of any sent media. This option is not enabled by default, but is commonly enabled to handle devices behind NAT. A change was made to the strict RTP support in the RTP stack to better tolerate late media when a reinvite occurs. When combined with the symmetric RTP support, this introduced an avenue where media could be hijacked. Instead of only learning a new address when expected, the new code allowed a new source address to be learned at all times. If a flood of RTP traffic was received, the strict RTP support would allow the new address to provide media, and (with symmetric RTP enabled) outgoing traffic would be sent to this new address, allowing the media to be hijacked. Provided the attacker continued to send traffic, they would continue to receive traffic as well. |
| CVE-2017-14098 | In the pjsip channel driver (res_pjsip) in Asterisk 13.x before 13.17.1 and 14.x before 14.6.1, a carefully crafted tel URI in a From, To, or Contact header could cause Asterisk to crash. |
| CVE-2016-9938 | 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. |
| CVE-2016-9937 | An issue was discovered in Asterisk Open Source 13.12.x and 13.13.x before 13.13.1 and 14.x before 14.2.1. If an SDP offer or answer is received with the Opus codec and with the format parameters separated using a space the code responsible for parsing will recursively call itself until it crashes. This occurs as the code does not properly handle spaces separating the parameters. This does NOT require the endpoint to have Opus configured in Asterisk. This also does not require the endpoint to be authenticated. If guest is enabled for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still processed and the crash occurs. |
| CVE-2015-2003 | The PJSIP PJSUA2 SDK before SVN Changeset 51322 for Android might allow attackers to execute arbitrary code by leveraging a finalize method in a Serializable class that improperly passes an attacker-controlled pointer to a native function. |
| CVE-2015-1558 | Asterisk Open Source 12.x before 12.8.1 and 13.x before 13.1.1, when using the PJSIP channel driver, does not properly reclaim RTP ports, which allows remote authenticated users to cause a denial of service (file descriptor consumption) via an SDP offer containing only incompatible codecs. |
| CVE-2014-8416 | Use-after-free vulnerability in the PJSIP channel driver in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1, when using the res_pjsip_refer module, allows remote attackers to cause a denial of service (crash) via an in-dialog INVITE with Replaces message, which triggers the channel to be hung up. |
| CVE-2014-8415 | Race condition in the chan_pjsip channel driver in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1 allows remote attackers to cause a denial of service (assertion failure and crash) via a cancel request for a SIP session with a queued action to (1) answer a session or (2) send ringing. |
| CVE-2014-8413 | The res_pjsip_acl module in Asterisk Open Source 12.x before 12.7.1 and 13.x before 13.0.1 does not properly create and load ACLs defined in pjsip.conf at startup, which allows remote attackers to bypass intended PJSIP ACL rules. |
| CVE-2014-6609 | The res_pjsip_pubsub module in Asterisk Open Source 12.x before 12.5.1 allows remote authenticated users to cause a denial of service (crash) via crafted headers in a SIP SUBSCRIBE request for an event package. |
| CVE-2014-4048 | The PJSIP Channel Driver in Asterisk Open Source before 12.3.1 allows remote attackers to cause a denial of service (deadlock) by terminating a subscription request before it is complete, which triggers a SIP transaction timeout. |
| CVE-2014-4045 | The Publish/Subscribe Framework in the PJSIP channel driver in Asterisk Open Source 12.x before 12.3.1, when sub_min_expiry is set to zero, allows remote attackers to cause a denial of service (assertion failure and crash) via an unsubscribe request when not subscribed to the device. |
| CVE-2014-2289 | res/res_pjsip_exten_state.c in the PJSIP channel driver in Asterisk Open Source 12.x before 12.1.0 allows remote authenticated users to cause a denial of service (crash) via a SUBSCRIBE request without any Accept headers, which triggers an invalid pointer dereference. |
| CVE-2014-2288 | The PJSIP channel driver in Asterisk Open Source 12.x before 12.1.1, when qualify_frequency "is enabled on an AOR and the remote SIP server challenges for authentication of the resulting OPTIONS request," allows remote attackers to cause a denial of service (crash) via a PJSIP endpoint that does not have an associated outgoing request. |
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