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2008-04-15Merged revisions 114148 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15Adding chanvar to SIPPEER from 1.4 branchOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15Shorten the mac address pattern, since some phones use different identifiers ↵Jason Parker
(such as the i2050 softphone). (closes issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff uploaded by c (license 460) Tested by: c_hans git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14Merged revisions 114120 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines The call_token on the pvt can occasionally be NULL, causing a crash. If it is NULL, we can skip this channel, since it can't the one we're looking for. (closes issue #9299) Reported by: vazir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14During hangup it is possible for p->chan or p->owner to be NULL, so just ↵Joshua Colp
return what the channel is bridged to instead of what they are *really* bridged to. Thanks Matt Nicholson! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14Merged revisions 114103 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines It is possible for the remote side to say they want T38 but not give any capabilities. (closes issue #12414) Reported by: MVF ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-12Make sure linkset is locked exiting ss7_start_callMatthew Fredrickson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-12Make sure we start incoming calls on SS7 with echo cancellation enabled. ↵Matthew Fredrickson
Also make sure when completing a COT we call ss7_start_call with the proper locks held. Lastly, make sure if we fail to get a channel from zt_new that we don't assume it's there. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-11Merged revisions 114083 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen. Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed. (issue #12400) Reported by: ztel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10A 'b' option has been added which causes chan_local to return the actual ↵Joshua Colp
channel that is behind it when queried. This is useful for transfer scenarios as the actual channel will be transferred, not the Local channel. If you have been using Local channels as queue members and having issues when the agent did a blind transfer this option may solve the issue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10Merged revisions 114045 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines Be sure that we're not about to set bridgepvt NULL prior to dereferencing it. (closes issue #11775) Reported by: fujin ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10Fix spelling of existent in a few places.Joshua Colp
(closes issue #12409) Reported by: candlerb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10Merged revisions 114021 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines Don't add custom URI options if they don't exist OR they are empty. (closes issue #12407) Reported by: homesick Patches: uri_options-1.4.diff uploaded by homesick (license 91) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09Merged revisions 113927 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines We need to set the persistant_route [sic] parameter for the sip_pvt during the initial INVITE, no matter if we're building the route set from an INVITE request or response. (closes issue #12391) Reported by: benjaminbohlmann Tested by: benjaminbohlmann ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09Enable enough RTP bridging to allow P2P to work.Joshua Colp
(closes issue #11901) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09Move all messages wrapped in skinnydebug from debug to verbose.Jason Parker
(closes issue #12224) Reported by: DEA Patches: chan_skinny-debug-log.txt uploaded by DEA (license 3) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09Merged revisions 113784 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario. (closes issue #12385) Reported by: viraptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09Merged revisions 113681 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a reinvite), then we should not send a BYE. (closes issue #12392) Reported by: fnordian Patches: chan_sip.patch uploaded by fnordian (license 110) with small modification from me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09Merged revisions 113596 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008) | 2 lines Initialize fr->cacheable to make valgrind happy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-08Merged revisions 113504 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr 2008) | 1 line Add a little more that is required for previously added devices. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-08Merged revisions 113454 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | 4 lines Add support for several new(ish) devices - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for providing me the required information. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-08Merged revisions 113348 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) | 7 lines Move check for still-bridged channels out a little further, to avoid possible deadlocks. (Closes issue #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt uploaded by Corydon76 (license 14) Tested by: callguy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07Merged revisions 113013 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/trunk ................ r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines Merged revisions 113012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07Merged revisions 113118 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines Allow playback with noanswer (and add earlyrtp option). (closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07Merged revisions 113012 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-05Found a little problem with the sip request handling that could lead to a ↵Steve Murphy
quick crash of asterisk, and a road to a DOS attack if left unfixed. Attaching to a running asterisk with "telnet hostname 5060", I would input "something", then hit return three times, and asterisk crashes. I traced it to handle_request_do(), which zeroes out the data (an ast_str ptr) if the string is too short. Instead of freeing the struct and nulling the pointer, it now just resets it, because this ast_str is expected by the calling routine to still be there after handle_request_do() returns. This appears to fix the crash. I assume that it was introduced with ast_str's being adopted. It's a subtle and easy-to-miss sort of problem. I also found all the places where the req.data is freed, and made sure the ptr is Nulled out as well; no good leaving bad ptrs laying around-- I didn't need to do this, but it seemed a good thing to do... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-04Merged revisions 112820 via svnmerge from Philippe Sultan
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line Free newly allocated channel before returning ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-04Merged revisions 112766 via svnmerge from Philippe Sultan
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines Prevent call connections when codecs don't match. (closes issue #10604) Reported by: keepitcool Patches: branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested by: phsultan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-03Merged revisions 112599 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines Fix the testing of the "res" variable so that it is more logically correct and makes the correct warning and debug messages print. (closes issue #12361) Reported by: one47 Patches: chan_zap_deferred_digit.patch uploaded by one47 (license 23) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-02Make MISDN generate channel rename events when the name changes.Tilghman Lesher
(closes issue #11142) Reported by: julianjm Patches: chan_misdn_tmpchan_trunk_v1.diff uploaded by julianjm (license 99) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-02Since the SIP request structure gets reused multiple times with TCP handling ↵Joshua Colp
we have to clear the debug state or else we will keep spitting out debug even after it has been turned off. (closes issue #12169) Reported by: pj Patches: 12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01Added dnsmgr status output for sip show registry.Jeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01Fix a typo that prevented configuration of non-dynamic peers.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01Existing DNS manager lookups extended to check for SRV records.Jeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01Fix last commitTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01This adds DNS SRV record support to DNS manager. If there is a SRV record ↵Jeff Peeler
for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01Merged revisions 112204 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered. (closes issue #11823) Reported by: SDamm ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01Demote a log message down to a warning.Joshua Colp
(closes issue #12345) Reported by: caio1982 Patches: limit_msg.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01Now that zaptel trunk has been removed, add the PSTN deprecation notice to ↵Russell Bryant
chan_zap, as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-31I missed a place when this define was changed.Jason Parker
(closes issue #12334) Reported by: ovi Patches: 12334-asterisk.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-31This fixes a high fence violation that MALLOC_DEBUG reported to me.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28This time the fix is proper for issue 12284. I have tested it thoroughly and ↵Mark Michelson
found that valgrind no longer complains and that calls do complete correctly. The fix is along the same lines as before: Make sure the final null terminator gets copied into the new sip_request's data pointer. Without it, parse_request will read and potentially write past the end of the string, causing potential crashes. (closes issue #12284...for real this time!) reported by falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28Temporary revert of 111662. It's causing lots of trouble and appears to not beMark Michelson
the proper solution to the problem reported anyway. (related to issue #12884) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28Merged revisions 111720 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar 2008) | 1 line Remove unimplemented softkeys. Prompted by issue #12325. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28The copy_request function did not take into account the necessary null ↵Mark Michelson
terminator for the string to be copied into. This resulted in parse_request reading invalid memory beyond the end of the string, and in some cases led to crashes. Thanks to falves11 for providing the valgrind output which led to the closure of this issue. (closes issue #12284) Reported by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26Oops, missed oneTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26Add expiry value to the sip show subscriptions CLI command.Joshua Colp
(closes issue #12025) Reported by: agx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26Large cleanup of DSP codeJason Parker
Per comments from dimas: 1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones. 2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before. 3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best. Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too. 4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used. 5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel. This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality. (closes issue #11968) Reported by: dimas Patches: v2-11968-dsp.patch uploaded by dimas (license 88) v4-11968-zap.patch uploaded by dimas (license 88) Tested by: dimas, qwell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26Merged revisions 111020 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be. (closes issue #11995) Reported by: fall ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26Merged revisions 110628 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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