|Tags||communication conferencing telephony sip|
14.5.031 May 2017 23:45 minor feature:
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.
14.4.015 Apr 2017 02:05 minor feature:
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.
14.3.014 Feb 2017 16:45 minor feature:
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.
14.2.025 Nov 2016 07:45 minor feature:
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.
14.1.027 Oct 2016 01:25 minor feature:
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.
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