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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines
Add a note about IAX2 to UPGRADE.txt.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines
Add IAX2 security document related to AST-2009-006.
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(closes issue #12912)
Reported by: rathaus
Tested by: tilghman, russell, dvossel, dbrooks
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From section 13.3.1.1 of RFC 3261:
If the UAS desires an extended period of time to answer the INVITE,
it will need to ask for an "extension" in order to prevent proxies
from canceling the transaction. A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction. To prevent cancellation, the UAS MUST send a non-100
provisional response at every minute, to handle the possibility of
lost provisional responses.
(closes issue #11157)
Reported by: rjain
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/315/
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In general channel names are in the form Foo/Bar-Z, but the channel name
could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to
truncate the channel name at the last hyphen.
(closes issue #15810)
Reported by: dhubbard
Patches:
dw-softhangup-1.4.patch uploaded by dhubbard (license 733)
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(closes issue #15787)
Reported by: tim_ringenbach
Patches:
chan_local.diff uploaded by tim ringenbach (license 540)
Tested by: tim_ringenbach
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We have kept this comment around long enough, that it's pretty clear that we're
keeping the code, because changing the code would require a pretty fundamental
architectural shift. We've also taken criticism in some quarters, because it
was believed that it was referring to the code being nasty. No, the code isn't
nasty, just the operation itself is rather odd. Fixed for eternity (probably
not).
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something we didn't allow before.
(closes issue #15714)
Reported by: pprindeville
Patches:
20090813__issue15714.diff.txt uploaded by tilghman (license 14)
Tested by: pprindeville
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have better information.
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(Figured out while working with issue #14906)
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In ast_write(), if a channel has a list of audiohooks, those
lists are written to and the resulting frame is what ast_write()
should continue with. The problem was the returned audiohook frame
was not being handled at all, and the original frame passed
into it did not contain the mixed audio, so essentially audio
was being lost. One result of this was chan_spy's whisper
mode no longer worked. To complicate the issue, frames
passed into ast_write may either be a single frame, or a list
of frames. So, as the list of frames is processed in the
audiohook_write, the returned frames had to be added to a new
list.
(closes issue #15660)
Reported by: corruptor
Tested by: dvossel
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(closes issue #15273)
Reported by: Benjamin Kluck
Patches:
say_c.patch uploaded by Benjamin Kluck (license 803)
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parethesis.
(closes issue #15242)
Reported by: Nick_Lewis
Patches:
pbx.c-funcparenthesis.patch2 uploaded by dbrooks (license 790)
pbx.c-funcparenthesis-1.4.diff uploaded by loloski (license 68)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@213970 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14730)
Reported by: pkempgen
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enabled.
(closes issue #15373)
Reported by: dcolombo
Patches:
chan_sip.patch uploaded by mbrancaleoni (license 342)
Tested by: dcolombo, mbrancaleoni
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(closes issue #15698)
Reported by: slavon
Patches:
20090817__issue15698.diff.txt uploaded by tilghman (license 14)
Tested by: slavon, tilghman
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dialplan.
(closes issue #15755)
Reported by: trendboy
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(closes issue #15751)
Reported by: atis
Patches:
ast_bridge_call_peer_cdr.patch uploaded by atis (license 242)
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(closes issue #15699)
Reported by: edantie
Patches:
mixmonitor.patch uploaded by edantie (license 862)
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to handle.
Without this patch, asterisk creates a temporary file before determining if the
specified command is valid. If invalid, we weren't properly cleaning up the file.
(closes issue #15730)
Reported by: zmehmood
Patches:
M15730.diff uploaded by junky (license 177)
Tested by: zmehmood
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If more ports were specified than configured in misdn.conf a reload would crash
asterisk. The problem was the unconfigured port was using data from the
previously configured port. When the data for an unconfigured port was freed a
crash would result from the double free.
(closes issue #12113)
Reported by: agupta
Patches:
bug12113.patch uploaded by jpeeler (license 325)
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The additional checks prevent generation of false TRANSFER events in certain situations.
(closes issue #14536)
Reported by: aragon
Patches:
queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
Tested by: aragon, mnicholson
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@211953 f38db490-d61c-443f-a65b-d21fe96a405b
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INVITEs.
There already was code present to be sure that a CANCEL will contain the same branch-id
as the INVITE it is cancelling. However, for INVITES which are challenged downstream,
this mechanism did not work properly. Now this is taken care of.
This is a backport of a fix already present in all 1.6.X branches and in trunk. It also
fixes ABE-1907.
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(ABE-1936)
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This is a partial revert of revision 82590, which was an attempted cleanup,
but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended
as a method by which component interfaces could be queried from the queue.
Membername isn't useful here, because that field cannot be used to obtain
further information about the member. See the documentation on
QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various
AMI commands which take a member argument for further justification.
(closes issue #15664)
Reported by: rain
Patches:
app_queue-queue_member_list.diff uploaded by rain (license 327)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@211038 f38db490-d61c-443f-a65b-d21fe96a405b
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we must lock the channel prior to modifying it.
(closes issue #15397)
Reported by: caspy
Patches:
20090714__issue15397.diff.txt uploaded by tilghman (license 14)
Tested by: caspy
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@210913 f38db490-d61c-443f-a65b-d21fe96a405b
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* Issue 15655: For the case where dialing is complete for an incoming
call, dahdi_new() was asked to start the PBX and then the code set more
channel variables. If the dialplan hungup before these channel variables
got set, asterisk would likely crash.
* Fixed potential for overlap incoming call to erroneously set channel
variables as global dialplan variables if the ast_channel structure failed
to get allocated.
* Added missing set of CALLINGSUBADDR in the dialing is complete case.
(closes issue #15655)
Reported by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@210575 f38db490-d61c-443f-a65b-d21fe96a405b
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Updated the imapstorage.txt documentation to reflect that issues with
c-client versions older than 2007 seem to cause crashing issues that
are not seen with more recent versions. Documentation has been updated
to reflect this.
(closes issue #14496)
Reported by: vbcrlfuser
Patches:
__20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, mmichelson, dbrooks
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Ensure that system headers located in /usr/local/include are actually treated
as system headers by the compiler, and not as local headers which are subject
to warnings from the -Wundef compiler option and others.
(closes issue #15606)
Reported by: mvanbaak
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Prior to this patch, a wildcard extension in the dialplan (for example, _*.) would take
precedence over picking up a call in the channel's pickup group. This patch simply moves
the block of code handling pickup group matching to above the extension matching code.
(closes issue #14735)
Reported by: stevedavies
Review: https://reviewboard.asterisk.org/r/319/
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developer discussions.
(related to issue #15639)
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(Related to issue #15639)
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(closes issue #15639)
Reported by: nmav
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(issue #15396)
Reported by: aragon
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@209879 f38db490-d61c-443f-a65b-d21fe96a405b
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When Milliwatt() was changed internally to use Playtones() so that the proper
tone was used, it introduced a drop in gain in the output signal. So, use
the playtones API directly and specify a volume argument such that the output
matches the gain of the original Milliwatt() code.
(closes issue #15386)
Reported by: rue_mohr
Patches:
issue_15386.rev2.diff uploaded by russell (license 2)
Tested by: rue_mohr
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@209838 f38db490-d61c-443f-a65b-d21fe96a405b
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The latest GCC (what will become 4.5.x) has a few new warnings, that in these
cases found some either downright buggy code, or at least seriously poorly
designed code that could be improved.
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There are some VoIP providers out there that will not accept SDP
offers with odd numbered UDPTL ports. While it is my personal opinion
that these VoIP providers are misinterpreting RFC 2327, it really is
not a big deal to play along with their silly little games. Of course,
since restricting UDPTL ports to only even numbers reduces the range
of available ports by half, so the option to use only even port numbers
is off by default. A user can enable the behavior by setting
use_even_ports=yes in udptl.conf.
(closes issue #15182)
Reported by: CGMChris
Patches:
15182.patch uploaded by mmichelson (license 60)
Tested by: CGMChris
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@209131 f38db490-d61c-443f-a65b-d21fe96a405b
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pthread_self returns a pthread_t which is not an unsigned int on all
pthread implementations. Casting it to an unsigned int fixes compiler warnings.
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Mostly trivial changes, but I did not know of any other way to fix the
"dereferencing type-punned pointer will break strict-aliasing rules" error
without creating a tmp variable in chan_skinny.
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I know what some of you are thinking: "UGH! Mark, why are you using
ast_strdup and ast_free for the string when you can just use ast_strdupa
and let the memory free itself?! Have the bats been chewing on your brain
again?"
Based on past experiences, I don't like using ast_strdupa inside a loop.
It's a good way to potentially exhaust stack space. Also, since this only
happens when reloading queues, I don't think that heap allocations and
frees are going to be a huge problem.
(closes issue #15559)
Reported by: amorsen
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