asterisk

Commit Graph

Author	SHA1	Message	Date
Brett Bryant	085b7b212a	Merged revisions 318919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318919 \| bbryant \| 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) \| 10 lines This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too much time has passed between sending audio. (closes issue #18206) Reported by: bernhardsi Patches: res_srtp_unhold.patch uploaded by bernhards (license 1138) Tested by: bernhards, notthematrix ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Andrew Latham	9f1a17f137	Replacing doc/* with wiki links Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	9653b5d500	Merged revisions 292309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292309 \| twilson \| 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) \| 10 lines Add sip show peer info about crypto and remove dated comment This patch adds information about the encryption setting to 'sip show peers' and removes an out-of-date comment from res_srtp.c and instead directs users to the proper documentation. (closes issue #18140) Reported by: chodorenko ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c81da53206	Merged revisions 292016 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292016 \| twilson \| 2010-10-15 16:40:56 -0500 (Fri, 15 Oct 2010) \| 5 lines Ref/unref res_srtp when we create/destroy a session This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp tries to unload before chan_sip does. Thanks, Russell! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	a51ce289b2	Merged revisions 287056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287056 \| twilson \| 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) \| 10 lines Don't hang up a call on an SRTP unprotect failure Also make it more obvious when there is an issue en/decrypting. (closes issue #17563) Reported by: Alexcr Patches: res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by: twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	920f5ea8b7	Merged revisions 284477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284477 \| twilson \| 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) \| 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP lines Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer. This patch caches the SRTP policies the we use so that we can change the ssrc and inform libsrtp of the new streams. It also uses the first acceptable a=crypto line from the incoming INVITE. (closes issue #17563) Reported by: Alexcr Patches: srtp.diff uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/878/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	857814f435	Add SRTP support for Asterisk After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

8 Commits (c98447f82c68be74b0987af38cef642db8ac24f2)