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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines
(closes issue #12694)
Reported by: yraber
Patches:
12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav
Thanks to file (Joshua Colp) for his IAX fix.
the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.
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until the originating line hangs up.
(closes issue #12382)
Reported by: one47
Patches:
zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23)
zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463)
Tested by: fleed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines
Don't try to send a response on a NULL pvt.
(closes issue #13919)
Reported by: barthpbx
Patches:
chan_iax2.c.patch uploaded by eliel (license 64)
Tested by: barthpbx
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r159246 | tilghman | 2008-11-25 15:40:28 -0600 (Tue, 25 Nov 2008) | 14 lines
Merged revisions 159245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) | 7 lines
Regression fix for last security fix. Set the iseqno correctly.
(closes issue #13918)
Reported by: ffloimair
Patches:
20081119__bug13918.diff.txt uploaded by Corydon76 (license 14)
Tested by: ffloimair
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) | 3 lines
System call ioperm is non-portable, so check for its existence in autoconf.
(Closes issue #13863)
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it would be best to maintain API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to ao2_callback() except
that it allows you to pass arbitrary data to the callback.
Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
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is only found if it dialed the extension that was subscribed to. You can now
specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as
it's value implies, ignore the current context of the caller when doing the
lookup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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our sip_pvt pointer in directly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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failed.
Without this reorder tone will play after hangup and both wedhorn's and my wife have threatened to use an axe on our asterisk box
(closes issue #13948)
Reported by: wedhorn
Patches:
switch.diff uploaded by wedhorn (license 30)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(issue #13948)
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(issue #13948)
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registration message pending.
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for this sscanf statement. Thanks Richard!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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on 32-bit systems, so we need to use %llu instead. Of course
%llu is 128-bits on 64-bit systems, so we have to cast to
unsigned long long. No harm, but it's sure annoying.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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from an int to a uint64_t. This prevents potential comparison
problems from happening if the version string exceeds
INT_MAX. This was an apparent problem for one user who could
not properly place a call on hold since the version in the
SDP of the re-INVITE to place the call on hold greatly
exceeded INT_MAX.
This also aligns with RFC 2327 better since it recommends
using an NTP timestamp for the version (which is a
64-bit number).
(closes issue #13531)
Reported by: sgofferj
Patches:
13531.patch uploaded by putnopvut (license 60)
Tested by: sgofferj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/trunk
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r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines
Begin on a crusade to end trailing whitespace!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines
We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
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trying to parse an invalid config
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines
Add some missing invite state changes necessary in the sip_write
function. Not setting the invite state correctly on the call was
resulting in the Record application leaving empty files. I also
have updated the doxygen comment next to the declaration of the
INV_EARLY_MEDIA constant to reflect that we also use this state
when we *send* a 18X response to an INVITE.
(closes issue #13878)
Reported by: nahuelgreco
Patches:
sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162)
Tested by: putnopvut
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changed from using a global lock in update_call_counter to
using the locks within the sip_pvt and sip_peer structures
instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Revert logic to mirror 1.4's in the sense that it will not allow
the call counter to dip below 0.
These two measures prevent potential races that could cause a SIP peer
to appear to be busy forever.
(closes issue #13668)
Reported by: mjc
Patches:
hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines
Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.
This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.
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the comparison is case insensitive.
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comparison
function is case insensitive.
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removed a very important line to set the "len" field
for incoming SIP requests. The result was that all incoming
SIP messages appeared to be 0-length, meaning Asterisk
could do no meaningful processing of anything SIP-related
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we get a GRS.
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'pri set debug' was changed from 'pri debug'.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) | 11 lines
Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not
to be sent, and instead, schedule a task to destroy the iax2 pvt structure
10 seconds later. This allows the IAX2 HANGUP packet to be queued,
transmitted, and ACKed before the pvt is destroyed.
(closes issue #13645)
Reported by: dzajro
Patches:
20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
Tested by: vazir
Reviewed: http://reviewboard.digium.com/r/51/
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- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines
Channel drivers assume that when their indicate callback
is invoked, that the channel on which the callback was called
is locked. This patch corrects an instance in chan_agent where
a channel's indicate callback is called directly without first
locking the channel.
This was leading to some observed locking issues in chan_local,
but considering that all channel drivers operate under the
same expectations, the generic fix in chan_agent is the right
way to go.
AST-126
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introduced
ast_channel_search_locked need to be made. Specifically, the caller needs to be able to
pass arbitrary data which in turn is passed to the callback. This patch addresses all
of the nested functions currently in asterisk trunk.
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- fix a spelling error in a header file
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that
the buffer is terminated.
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makes more sense. Where it was set before, it was impossible
to actually delay sending a CANCEL if we had not yet received
a provisional response to an INVITE.
(closes issue #13626)
Reported by: atis
Patches:
13626.patch uploaded by putnopvut (license 60)
Tested by: atis
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ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either
of these mandates that the passed 'arg' is a hashable object, making
searching for an ao2 object based on outside criteria difficult.
Reviewed by Russell and Mark M. via ReviewBoard:
http://reviewboard.digium.com/r/36/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines
Clarify error message.
(closes issue #13809)
Reported by: denke
Patches:
20081104__bug13809.diff.txt uploaded by Corydon76 (license 14)
Tested by: denke
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invalid extension is called.
(closes issue #13408)
Reported by: adomjan
Patches:
chan_dahdi.c-ss7-unallocated-2 uploaded by adomjan (license 487)
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from pre-astobj2ified chan_sip.
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container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would
only remove a single object.
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added at the same time as progress_with_cause
move parsing of the qsig channel mapping configuration option outside ifdef HAVE_PRI_INBANDDISCONNECT and into a properly ifdef'd block
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pri_progress_with_cause function to have libpri support work.
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AGENT() function XML documentation.
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channel list calling a caller-defined callback. The callback returns non-zero
if a match is found. This should speed up some of the code that I committed
earlier today in chan_sip (which is also updated by this commit).
Reviewed by russellb and kpfleming via ReviewBoard:
http://reviewboard.digium.com/r/28/
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