querying realtime from the dialplan a little more consistent and easy to use.
The original REALTIME function is preserved, for those who are already
accustomed to that interface.
(closes issue #13651)
Reported by: Corydon76
Patches:
20081119__bug13651__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage, Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This provides a new timing interface. In order to use it,
you must be running a Linux with a kernel version of
2.6.25 or newer and glibc 2.8 or newer.
This timing interface is a good alternative if a timing
source is necessary (e.g. for IAX trunking) but DAHDI is
otherwise unnecessary for the system.
For now, this commit contains the actual work done in the
res_timing_timerfd branch. There are no notices in the README
or CHANGES files yet, but they will be added in my next commit.
The timing API of Asterisk also needs to have a bit of work done
with regards to choosing which timing interface to use. This commit
makes the choice a build-time decision, by only allowing one of
the timer interfaces to be chosen in menuselect. It would be preferable
if the choice could be made at run-time, however. The preferred timing
interface could be loaded and tested, and if it does not work, choice
number two may be used instead. That sort of thing. That is beyond
the scope of work in this branch though.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
in 1.6.1 (as compared to 1.6.0 and versions of 1.4), this change also
deprecates the use of Asterisk with FreeBSD 4, given the central use of va_copy
in core functions. va_copy() is C99, anyway, and we already require C99 for
other purposes, so this isn't really a big change anyway. This change also
simplifies some of the core ast_str_* functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
is unloaded and then Asterisk is stopped. The problem was that
we are not unregistering the ast_moh_destroy function at exit.
(closes issue #13761)
Reported by: eliel
Patches:
res_musiconhold.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
is in use so that the message is deleted from both
local and IMAP storage.
(closes issue #13642)
Reported by: jaroth
Patches:
deleteyes.patch uploaded by jaroth (license 50)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
* 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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines
(closes issue #13899)
Reported by: akkornel
This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: alex70
Patches:
13420.13539.patch uploaded by murf (license 17)
Tested by: murf, awk
This fixes two problems: a spurious linefeed insertion
probably left over from pre-precomment times. Only
generated when category had no previous comments.
The other problem: Insertions could get the line-numbering
out of whack and generate negative line numbers, causing
chunks of line numbers to be emitted, on the scale of the
number of lines up to that point in the file. In such cases,
abort the looping, and all is well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a function: in these cases, we have to use the heap, or garbage will result.
(closes issue #13898)
Reported by: alecdavis
Patches:
20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov 2008) | 1 line
dist-clean should remove dependency information files as well
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r157163 | kpfleming | 2008-11-15 20:31:03 +0100 (Sat, 15 Nov 2008) | 1 line
when an individual directory dist-clean is run, run clean in that directory first, and when running top-level dist-clean, do not run subdirectory clean operations twice
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157104 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov 2008) | 13 lines
major update to doxygen configuration file:
1) update to doxygen 1.5.x style file, as used in trunk
2) tell doxygen where are header files are, so include-file processing can be done
3) make all macros that are used to define variables/functions be expanded, so that doxygen will properly document the resulting variable/function
4) make all macros that are used to provide the contents of a variable (structure) be expanded, so that doxygen will be able to document the resulting fields
5) suppress compiler attributes (__attribute__(xxx)) from being seen by doxygen, so it will properly match up function definition and usage (for an example of th effect of this, look at the doxygen docs for ast_log() from before and afte this commit)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
hashing used by app_queue.c to be case-insensitive.
This is accomplished by adding a new case-insensitive
hashing function.
This was necessary to prevent bad refcount errors
(and potential crashes) which would occur due to the
fact that queues were initially read from the config
file in a case-sensitive manner. Then, when a user
issued a CLI command or manager action, we allowed
for case-insensitive input and used that input to
directly try to find the queue in the hash table. The result
was either that we could not find a queue that was input or
worse, we would end up hashing to a completely bogus value
based on the input.
This commit resolves the problem presented in
issue #13703. However, that issue was reported against
1.6.0. Since this fix introduces a behavior change, I am
electing to not place this same fix in to the 1.6.0 or 1.6.1
branches, and instead will opt for a change which does not
change behavior.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines
If the prompt to reenter a voicemail password timed out, it
resulted in the password not being saved, even if the input matched
what you gave when first prompted to enter a new password. This is
because the return value of ast_readstring was checked, but not checked
properly.
This bug was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines
ast_waitfordigit() requires that the channel be up, for no good logical
reason. This prevents While/EndWhile from working within the "h"
extension.
Reported by: jgalarneau (for ABE C.2)
Fixed by: me
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: smurfix
This reverts a change I made in 116297. At the time it seemed the change was required to solve an issue with attempting a transfer but then letting it timeout without dialing any digits. However, I didn't realize that having an empty extension was possible. I'm removing the immediate return that was added in pbx_find_extension if the extension is null.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- MeetMe()
- MeetMeCount()
- MeetMeChannelAdmin()
- MeetMeAdmin()
- SLAStation()
- SLATrunk()
- Add an attribute to optionlist 'hasparams' with the same functionality as the one
we have in <parameter> and <argument> (the DTD was updated)
- Fix a leak when getting an attribute while parsing an <optionlist>.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Mark Michelson
Tilghman Lesher
Terry Wilson
Kevin P. Fleming
Sean Bright
Jason Parker
Jeff Peeler
Steve Murphy
Eliel C. Sardanons
Russell Bryant
Matthew Fredrickson