asterisk 14.7.2

Asterisk is an open source framework for building communications applications. Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions. It is used by small businesses, large businesses, call centers, carriers and government agencies, worldwide. Asterisk is free and open source.

Tags communication conferencing telephony sip
License GNU GPL
State stable

Recent Releases

14.7.218 Nov 2017 00:05 minor feature: Res_pjsip: Add to list of valid characters for from_user. a regression where some characters were unable to be used in. The from_user field of an endpoint. Additionally, the backtick was Removed from the list of valid characters, since it is not valid, And it was replaced with a single quote, which is a valid character. res_pjsip_registrar.c: AOR and pjproject group deadlock. One of the patches for ASTERISK_27147 introduced a deadlock regression. When the connection oriented transport shut down, the code attempted to. Remove the associated contact. However, that same transport had just Requested a registration that we hadn't responded to yet. Depending Upon timing we could deadlock. Made send the REGISTER response after we completed processing the. Request contacts and released the AOR lock to avoid the deadlock.
14.7.109 Nov 2017 14:25 minor feature: AST-2017-009: pjproject: Add validation of numeric header values Parsing the numeric header fields 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. To address this, new "strto" functions Have been created that do range checking and those functions are Used wherever possible in the parser. Created pjlib/include/limits.h and pjlib/include/compat/limits.h to either include the system limits.h or define common numeric. Limits if there is no system limits.h. Created strto*_validate functions in sip_parser that take bounds. And on failure call the on_str_parse_error function which prints an error message and calls PJ_THROW. Updated sip_parser to validate the numeric fields. an in sip_transport that prevented error messages. From being properly displayed. Added "volatile" to some variables referenced in PJ_CATCH blocks as the optimizer was sometimes optimizing them away. Length calculation in sip_transaction/create_tsx_key_2543 to account for signed ints being 11 characters, not 9. AST-2017-011 - res_pjsip_session: session leak when a call is rejected. A previous commit made it so when an invite session transitioned into a. Disconnected state destruction of the Asterisk pjsip session object was Postponed until either a transport error occurred or the event timer Expired. However, if a call was rejected (for instance a 488) before the Session was fully established the event timer may not have been initiated, or it was canceled without triggering either of the session finalizing states. Mentioned above. Really the only time destruction of the session should be delayed is when a BYE is being transacted. This is because it's possible in some cases for the. Session to be disconnected, but the BYE is still transacting. This patch makes it so the session object always gets released (no more. Memory leak) when the pjsip session is in a dis
14.6.220 Sep 2017 18:05 minor feature: AST-2017-008: Improve RTP and RTCP packet processing. Validate RTCP packets before processing them. Validate that the received packet is of a minimum length and apply the RFC3550 RTCP packet validation checks. Potentially reading garbage beyond the received RTCP record data. Rtp- themssrc only being set once when the remote could change. The SSRC. We would effectively stop handling the RTCP statistic records. Rtp- themssrc to not treat a zero value as special by adding. Rtp- themssrc_valid to indicate if rtp- themssrc is available. strict RTP learning from always accepting the first new address Packet as the new stream. Strict RTP to initialize the expected sequence number with the. Last received sequence number instead of the last transmitted sequence Number. The predicted next sequence number calculation in. Rtp_learning_rtp_seq_update() to handle overflow.
14.6.102 Sep 2017 06:45 minor feature: Pjsip_message_ip_updater: handling "tel" URIs Sanitize_tdata was assuming all URIs were SIP URIs so when a non SIP uri was in the From, To or Contact headers, the unconditional. Cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused a segfault when trying to access uri- other_param. Added PJSIP_URI_SCHEME_IS_SIP(uri) PJSIP_URI_SCHEME_IS_SIPS(uri). Checks before attempting to cast or use the returned uri. AST-2017-006: app_minivm application MinivmNotify command injection An admin can configure app_minivm with an externnotify program to be run. When a voicemail is received. The app_minivm application MinivmNotify Uses ast_safe_system() for this purpose which is vulnerable to command Injection since the Caller-ID name and number values given to externnotify Can come from an external untrusted source. Add ast_safe_execvp() function. This gives modules the ability to run. External commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new. Function does not allow malicious input to break argument encoding. This May be of particular concern where CALLERID(name) or CALLERID(num) may be Used as a parameter to a script run by ast_safe_system() which could Potentially allow arbitrary command execution. Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp(). Instead of ast_safe_system() to avoid command injection. Document code injection potential from untrusted data sources for other. Shell commands that are under user control. res_rtp_asterisk: Only learn a new source in learn state. This change moves the logic which learns a new source address. For RTP so it only occurs in the learning state. The learning State is entered on initial allocation of RTP or if we are Told that the remote address for the media has changed. While in the learning state if we continue to receive media from. The original source we restart the learning process. It is Only once we receive a sufficien
14.4.121 May 2017 12:25 minor feature: AST-2017-003: Handle zero-length body parts correctly. AST-2017-004: chan_skinny: Add EOF check in skinny_session The while(1) loop in skinny_session wasn't checking for EOF so a packet that was longer than a header but still truncated. Would spin the while loop infinitely. Not only does this Permanently tie up a thread and drive a core to 100 utilization, The call of ast_log() in such a tight loop eats all available Process memory. Added poll with timeout to top of read loop. AST-2017-002: Ensure transaction key buffer is large enough.
14.3.106 Apr 2017 06:45 minor feature: CDR: Protect from data overflow in ast_cdr_setuserfield. Ast_cdr_setuserfield wrote to a length field using strcpy. This could. Result in a buffer overrun when called from chan_sip or func_cdr. This patch Adds a maximum bytes written to the field by using ast_copy_string instead.
14.2.109 Dec 2016 23:45 minor feature: Update for 14.2.1 chan_sip: Do not allow non-SP/HTAB between header key and colon. RFC says SIP headers look like: HCOLON = *( SP / HTAB ) ":" SWS SWS = LWS ; sep whitespace LWS = *WSP CRLF 1*WSP ; linear whitespace WSP = SP / HTAB ; from rfc2234. chan_sip implemented this: HCOLON = *( LOWCTL / SP ) ":" SWS LOWCTL = x00-1F ; CTL without DEL. This discrepancy meant that SIP proxies in front of Asterisk with chan_sip could pass on unknown headers with x00- x1F in them, which would be treated by Asterisk as a different (known) header. For example, the "To x01:" header would gladly be forwarded by some proxies as irrelevant, but chan_sip would treat it as the relevant "To:" header. Those relying on a SIP proxy to scrub certain headers could mistakenly get unexpected and unvalidated data fed to Asterisk. This change so chan_sip only considers SP/HTAB as valid tokens before the colon, making it agree on the headers with other speakers of SIP. res_format_attr_opus: crash when fmtp contains spaces. When an opus offer or answer was received that contained an fmtp line with spaces between the attributes the module would fail to properly parse it and crash due to recursion. This change makes the module handle the space properly and also removes the recursion requirement.
14.1.212 Nov 2016 10:05 minor feature: Revert "chan_sip: lastrtprx always updated" This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc. Unfortunately, the aforementioned commit caused a regression (incoming calls would eventually disconnect). Thus it is being removed.
14.1.103 Nov 2016 19:25 minor feature: App_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS. When executing the MailboxExists dialplan application and MAILBOX_EXISTS dialplan function the passed in temporary voice. Mailbox was not cleared, causing it to try to free garbage.
14.0.204 Oct 2016 03:15 minor feature: Release summaries: Remove previous versions version: Update for 14.0.2. lastclean: Update for 14.0.2. realtime: Add database scripts for 14.0.2. logger: Output early verbose messages to console. Verbose messages should be printed to the console if the sublevel is less than option_verbose. This ensures the welcome message with copyright and license are printed at daemon and interactive rasterisk startup. Remove "format_ogg_opus: New format". This reverts commit 40aa28131bc30b4516da2b20eb1a1e043920169c. download_externals: with re-install. Needed to ignore an xmlstarlet return code for optional element.
14.0.026 Sep 2016 20:45 major feature: asterisk 14.0.0 Released.