When calculating the 'A2' portion of our digest for verification, we need the
HTTP method that is currently in use. Unfortunately our mapping function was
incorrect, resulting in invalid hashes being generated and, in turn, failures
in authentication.
(closes issue #18598)
Reported by: ksn
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Parenthesis in the wrong position. Regression from issue #14365 when
expanding conference flags to use 64 bits.
(closes issue #18418)
Reported by: MrHanMan
Tested by: rmudgett
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r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) | 12 lines
Fixes session-timers=refuse not being enforced for *caller*
During handle_request_invite, the session timer mode was retrieved from
a cached variable. This patch forces a peer lookup of the session timer
mode in the case of an incoming invite.
(closes issue #18804)
Reported by: wdoekes
Patches:
issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717)
issue_18804_v2.diff uploaded by dvossel (license 671)
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r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May 2011) | 10 lines
Honor the C option to MeetMe when L is passed.
This fixes a case that r304773 and friends missed.
(closes issue #17317)
Reported by: var
Patches:
meetme-continue-on-l_16218.diff uploaded by var (license 1227)
Tested by: seanbright
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After dahdi_hangup() has supposedly hungup an ISDN channel there is still
traffic on the S0-bus because the channel was not cleaned up fully.
Shuffled the hangup code to include some missing cleanup. Also fixed some
code formatting in the area. I think the primary missing clean up code
was the call to tone_zone_play_tone() to turn off any active tones on the
channel.
(closes issue #19188)
Reported by: jg1234
Patches:
issue19188_v1.8.patch uploaded by rmudgett (license 664)
Tested by: jg1234
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This has already been discussed and should have been resolved earlier. View
revsion 285565's log for more information about why it is important to not
put timer in the Require header.
(closes issue #18704)
Reported by: mfrager
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines
Merged revisions 315891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines
Fix our compliance with RFC 3261 section 18.2.2.
This change optimizes the free_via() function and removes some redundant null
checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
the port specified in the Via header for routing responses (even when maddr is
not set). Also the htons() function is now used when setting the port.
Additional documentation comments have been added in various places to make the
logic in the code clearer.
(closes issue #18951)
Reported by: jmls
Patches:
issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)
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r315672 | twilson | 2011-04-26 15:52:25 -0700 (Tue, 26 Apr 2011) | 18 lines
Merged revisions 315671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) | 11 lines
Make sure unregistering a peer unlinks it from the peer container
Instead of mostly copying the code from expire_register, just use the function
that "does the right thing".
(closes issue #16033)
Reported by: kkm
Patches:
016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
Tested by: kkm, tilghman, twilson
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The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if
any of them do not exist. Many of the places the 'e' extension was
supposed to be invoked fail because the priority was set wrong. There
were two places where the 'e' extension was not even checked for fall
back.
* Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T'
extension check and added the 'e' extension as a fall back to the two
missing locations.
* Prioritized and optimized some hangup tests associated with the 'e'
extension.
(closes issue #19136)
Reported by: kshumard
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines
Merged revisions 315596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines
Allow transfer loops without allowing forwarding loops
We try to avoid the situation where two phones may be forwarded to each other
causing an infinite loop by storing each dialed interface in a channel
datastore and checking the list before dialing out. This works, but currently
breaks situations like A calls B, A transfers B to C, B transfers C to A, and A
transfers C to B. Since human interaction is happening here and not an
automated forwarding loop, it should be allowed.
This patch removes the dialed_interfaces datastore when a call is bridged (a
suggestion from the brilliant mmichelson). If a call is being bridged, it
should be safe to assume that we aren't stuck in a loop.
Since we are now handling this is the bridge code, the previous attempts at
handling it in app_dial and app_queue are removed.
Review: https://reviewboard.asterisk.org/r/1195/
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r315502 | tilghman | 2011-04-26 14:22:52 -0500 (Tue, 26 Apr 2011) | 21 lines
Merged revisions 315501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines
Fix the bounds-checking code.
The code that set the bit within the select bitfield was correct, but the
bounds-checking code was not. The change to that line uses the new _bitsize
macro for clarity. Also, FD_ZERO macro did not zero-out anything but the
first word of the bitfield, so this could have caused problems with modules
using that macro with the expanded bitfield.
(closes issue #18773)
Reported by: jamicque
Patches:
20110423__issue18773.diff.txt uploaded by tilghman (license 14)
Tested by: chris-mac
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch resolves a fairly complex deadlock that can occur with the
combination of chan_local and a dialplan switch, such as dynamic realtime
extensions, which pulls autoservice into the picture when doing a dialplan
lookup.
(closes issue #18818)
Reported by: nic
Patches:
issue18818.patch uploaded by jthurman (license 614)
18818.v1.txt uploaded by russell (license 2)
Tested by: nic, jthurman, kterzi, steve-howes, sysreq, IshMalik
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011) | 11 lines
Initialize buffers in getvar and getvarfull.
Initialize the buffers used to hold the result from GET VARIABLE or
GET VARIABLE FULL. The bug report shows func_read returning garbage in
the result. It assumed that the buffer passed in was initialized, like many
other functions do. In the more common code path (through the dialplan), it
is initialized, so just initialize it here too.
(closes issue #19050)
Reported by: johnz
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r314776 | russell | 2011-04-22 08:35:22 -0500 (Fri, 22 Apr 2011) | 10 lines
Fix handling of some call parking config options.
This patch adjusts the handling of some call parking config options to fix some
issues that have already been addressed in 1.8 and trunk.
(closes issue #19167)
Reported by: bluecrow76
Patches:
asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff uploaded by bluecrow76 (license 270)
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) | 6 lines
Don't allocate more space than necessary for a sip_pkt
This extra allocation is a hold-over from when pkt->data was a
character array. Now that it is an allocated string, just allocate
enough for the sip_pkt.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Return correct status: SUCCESS/FAILED/HANGUP. Previously, abnormal
exits from the command loop such as hangup would return SUCCESS.
* The "asyncagi break" command now returns SUCCESS and is now the only way
to break the command loop with that status. Previously, it returned
FAILED.
* The AMI event AsyncAGI End is no longer sent if the AsyncAGI Start event
is not sent. Previously, this happened because of an error setting up the
AGI pipes.
* All executed AGI commands now get an AsyncAGI Exec result event.
Previously, if the command returned failure (because of hangup), the
command loop just exited with FAILURE and did not send the AsyncAGI Exec
result event.
* Makes sure that the channel frame queue is empty on hangup.
Review: https://reviewboard.asterisk.org/r/1183/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
