Age | Commit message (Collapse) | Author | Files | Lines |
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
(closes issue #18693)
........
r310734 | may | 2011-03-15 00:45:53 +0300 (Tue, 15 Mar 2011) | 12 lines
Introduce t.38 parameters control functionality not full but enough for
Send/RcvFax support
Introduce t.38 controls between asterisk core and channel/proto layers.
Not all parameters are transferred from proto layers but *Fax apps
tested and work ok.
(issue #18693)
Reported by: benngard2
Patches:
issue-18693.patch uploaded by may213 (license 454)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310735 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r310636 | rmudgett | 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) | 39 lines
Merged revisions 310635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines
Merged revisions 310633 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines
"Caller*ID failed checksum" on Wildcard TDM2400P and TDM410
The last character in the caller id message is getting a framing error.
The checksum is the last character in the message. A framing error in the
checksum could be because:
1) The sender did not send a full stop bit.
2) The sender cut off the FSK carrier too soon.
3) The sender opted to send zero of the specified zero to 10 trailing mark
bits and round-off errors in the code resulted in the code not being where
it thought it was in the demodulated bit stream.
Bit 8 of 'b' is set when parity error.
Bit 9 of 'b' is set when framing error.
Made ignore the framing and parity error bits if the errored character is
the checksum. We can tolerate a framing/parity error there. The checksum
character validates the message.
(closes issue #18474)
Reported by: nivek
Patches:
callerid.c.1.patch uploaded by nivek (license 636) (with modifications)
Tested by: nivek
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310637 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r310587 | jrose | 2011-03-14 10:27:57 -0500 (Mon, 14 Mar 2011) | 15 lines
Merged revisions 310585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | 8 lines
Adds 'p' as an option to func_volume. When it is on, the old behavior with DTMF controlling volume adjustment will be enforced.
When it is off, DTMF will not be processed by the function.
Programmed by Jonathan Rose
Reviewed by David Vossel, Leif Madsen, and Russell Bryant
http://reviewboard.digium.internal/r/93/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310588 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
various OS distributions. Thanks David.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310547 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r310462 | tilghman | 2011-03-12 14:27:54 -0600 (Sat, 12 Mar 2011) | 45 lines
Merged revisions 310448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r310448 | tilghman | 2011-03-12 14:24:54 -0600 (Sat, 12 Mar 2011) | 38 lines
Recorded merge of revisions 310435 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011) | 31 lines
Add AELSub, which provides a stable entry point into AEL subroutines.
This commit needs some explanation, given that we're adding a new application
into an existing release branch. This is generally a violation of our release
policy, except in very limited circumstances, and I believe this is one of
those circumstances.
The problem that this solves is one of the sanity of using multiple dialplan
languages to define a dialplan. In the case of the reporter, he or she is
using AEL is define subroutines, while using Realtime extensions to invoke
those subroutines. While you can do this, it's based upon the reality of AEL
using actual dialplan extensions; however, there is no guarantee that the
details of _how_ AEL is compiled into extensions will remain stable. In fact,
at the time of this commit, it has already changed twice, once in a
fundamental way.
Now normally, a new application would only be added to trunk. However, this
application is explicitly to create a stable user-level API between versions,
and adding it to trunk only will not solve the user's problem of switching
between 1.6.2 and 1.8, nor will it help anybody switching from 1.8 to 1.10.
Therefore, it needs to go into existing release branches. For the sake of
consistency, and also because one of the changes was between 1.4 and 1.6.x,
I am also electing to commit this to 1.4.
(closes issue #18910)
Reported by: alexandrekeller
Patches:
20110304__issue18919__1.6.2.diff.txt uploaded by tilghman (license 14)
20110304__issue18919__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: alexandrekeller
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310500 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r310415 | tilghman | 2011-03-12 14:05:46 -0600 (Sat, 12 Mar 2011) | 14 lines
Merged revisions 310414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011) | 7 lines
Transactional handles should be used for the insertbuf, if available.
Also, fix a possible resource leak.
(closes issue #18943)
Reported by: irroot
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310416 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310373 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Recent versions of GCC have a tuning option value of 'native', which causes
the compiler to optimize the build for the CPU the compile is performed on.
Since most people are building Asterisk on the machine they plan to run it on,
the configure script and build system will now use this value unless a different
value is specified by the user in CFLAGS when the configure script is executed.
In addition, this value will be used for building the GSM and LPC10 codecs as
well, in preference to the logic that has been in their Makefiles forever to
optimize for certain types of CPUs.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310332 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines
remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call
If the channel condition is one of the following after breaking out of the loop, don't try to update_peer
(where x = 0/1)
1). ZOMBIE
2). cx->tech_pvt != pvtx
3). gluex != ast_rtp_instance_get_glue(cx->tech->type))
(closes issue #18781)
Reported by: alecdavis
Patches:
bug18781.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, ZX81
Review: https://reviewboard.asterisk.org/r/1128/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310288 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) | 13 lines
Add \r\n to remaining http headers passed to ast_http_send
r309204 changed the behavior of ast_http_send. It now requires headers
to be passed with trailing \r\n. This change updates the remaining
instances in the code that did not pass the \r\n.
(closes issue #18186)
Reported by: nivaldomjunior
Patches:
res_phoneprov.c.diff uploaded by lathama (license 1028)
manager.diff.txt uploaded by twilson (license 396)
Tested by: lathama
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310241 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar 2011) | 9 lines
Be more tolerant of what URI we accept for call completion PUBLISH requests.
(closes issue #18946)
Reported by: GeorgeKonopacki
Patches:
18946.patch uploaded by mmichelson (license 60)
Tested by: GeorgeKonopacki
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310238 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r310142 | tilghman | 2011-03-09 23:53:29 -0600 (Wed, 09 Mar 2011) | 19 lines
Merged revisions 310141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines
Merged revisions 310140 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines
Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems.
(closes issue #18295)
Reported by: pruiz
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310143 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) | 9 lines
Returns with an error notice if CHANNEL function of SIP channel is read without arguments.
(Closes issue #18653)
Reported by: wuwu
Patches:
diff.patch uploaded by jrose (license 1225)
Tested by: jrose
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310089 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r310039 | twilson | 2011-03-08 10:10:50 -0800 (Tue, 08 Mar 2011) | 11 lines
Spelling fix in "calendar show calendar"
s/Cartegories/Catagories/
(closes issue #18931)
Reported by: pdugas
Patches:
res_calendar.c.patch uploaded by pdugas (license 1222)
Review: [full review board URL with trailing slash]
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310045 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) | 1 line
Make pri parameter description consistent.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309996 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309858 | jrose | 2011-03-07 16:07:25 -0600 (Mon, 07 Mar 2011) | 22 lines
Merged revisions 309857 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r309857 | jrose | 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) | 15 lines
Merged revisions 309856 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | 8 lines
Bug fix for MixMonitor involving filenames with '.' not in the extension
Closes issue #18391)
Reported by: pabelanger
Patches:
bugfix.patch uploaded by jrose (license 1225)
Tested by: jrose
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309859 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines
Merged revisions 309251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines
Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround.
Not surprisingly, the workaround was exactly the same code as was provided by
the Flex maintainers, albeit in two different places, in different macros.
This should fix the FreeBSD builds, which have an older version of Flex.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309809 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, 06 Mar 2011) | 3 lines
Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309766 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) | 6 lines
Fix caller id passed to openr2_chan_make_call
(closes issue #18894)
Reported by: malufrj
Tested by: moy
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309721 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309678 | tilghman | 2011-03-05 04:29:30 -0600 (Sat, 05 Mar 2011) | 14 lines
Merged revisions 309677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) | 7 lines
Missed part of the conversion when we started passing ppid to astcanary.
(closes issue #18850)
Reported by: viraptor
Patches:
canary_ppid.patch uploaded by viraptor (license 543)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309679 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Adding the setvar option with variable substitution on the value allows things
like setting the outbound caller id name to the summary of a calendar event,
etc. Values could be chained together as they are appended in order to do some
scripting if necessary.
Review: https://reviewboard.asterisk.org/r/1134/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309640 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309585 | mnicholson | 2011-03-04 13:38:25 -0600 (Fri, 04 Mar 2011) | 9 lines
Merged revisions 309584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar 2011) | 2 lines
Restore mysterious lua_pushvalue() call removed in r309494. The mystery has been solved.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309587 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309542 | mnicholson | 2011-03-04 13:00:33 -0600 (Fri, 04 Mar 2011) | 11 lines
Merged revisions 309541 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar 2011) | 4 lines
Check for errors from fseek() when loading config file, properly abort on errors from fread(), and supply a traceback for errors generated when loading the config file.
Also, prepend a newline to traceback output so that the main error message is on it's own line.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309543 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309495 | mnicholson | 2011-03-04 12:10:23 -0600 (Fri, 04 Mar 2011) | 9 lines
Merged revisions 309494 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar 2011) | 2 lines
remove mysterious lua_pushvalue() that is never used
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309496 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #16024)
Reported by: mnicholson
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309493 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Asterisk).
In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309491 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309448 | mnicholson | 2011-03-04 09:59:25 -0600 (Fri, 04 Mar 2011) | 8 lines
Export global symbols from pbx_lua to allow modules to be loaded. Fixes a regression introduced in r278132.
(closes issue #18671)
Reported by: Igels
Patches:
pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96)
Tested by: Igels
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309449 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
Get real channel of a DAHDI call.
Starting with Asterisk v1.8, the DAHDI channel name format was changed for
ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
There were several reasons that the channel name had to change.
1) Call completion requires a device state for ISDN phones. The generic
device state uses the channel name.
2) Calls do not necessarily have B channels. Calls placed on hold by an
ISDN phone do not have B channels.
3) The B channel a call initially requests may not be the B channel the
call ultimately uses. Changes to the internal implementation of the
Asterisk master channel list caused deadlock problems for chan_dahdi if it
needed to change the channel name. Chan_dahdi no longer changes the
channel name.
4) DTMF attended transfers now work with ISDN phones because the channel
name is "dialable" like the chan_sip channel names.
For various reasons, some people need to know which B channel a DAHDI call
is using.
* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
in use by the channel. Use CHANNEL(no_media_path) to determine if the
channel even has a B channel.
* Added AMI event DAHDIChannel to associate a DAHDI channel with an
Asterisk channel so AMI applications can passively determine the B channel
currently in use. Calls with "no-media" as the DAHDIChannel do not have
an associated B channel. No-media calls are either on hold or
call-waiting.
(closes issue #17683)
Reported by: mrwho
Tested by: rmudgett
(closes issue #18603)
Reported by: arjankroon
Patches:
issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: stever28, rmudgett
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309446 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines
Merged revisions 309356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines
Merged revisions 309355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines
fix small memory leak
fix small memory leak caused by a string allocation that wasn't freed
(closes issue #18907)
Reported by: andy11
Patches:
asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224)
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309404 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #18226)
Reported by: clegall_proformatique
Patches:
asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309300 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines
Merged revisions 309255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines
Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP.
Since it's a duplicate, nothing is going to be done, so delme doesn't need to
be set at all. Strangely, when this was added, this was being set to 1 in 1.6,
and 0 in trunk.
(issue AST-439)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309257 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | 7 lines
Fix consistency of CRLFs on HTTP headers that get sent out.
(closes issue #18186)
Reported by: nivaldomjunior
Patches:
18186-httpheadernewline.diff uploaded by qwell (license 4)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309209 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 Mar 2011) | 7 lines
Document CHANNEL(keypad_digits) and CHANNEL(no_media_path).
* Added XML documentation for CHANNEL(keypad_digits) and
CHANNEL(no_media_path).
* Tweaked XML documentation for CHANNEL(reversecharge).
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309171 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) | 16 lines
Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal.
Looks like an unintended change when sig_analog.c was extracted from
chan_dahdi.c.
Removed useless conditional around needed code and fixed resulting
compiler warning.
(closes issue #18667)
Reported by: enegaard
Patches:
issue18667.patch uploaded by enegaard (license 1197)
Tested by: enegaard
JIRA SWP-2965
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309127 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309084 | dvossel | 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) | 15 lines
Merged revisions 309083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines
Fixes thread blocking issue in the sip TCP/TLS implementation.
(closes issue #18497)
Reported by: vois
Patches:
issues_18497.diff uploaded by dvossel (license 671)
Tested by: vois, rossbeer, kowalma, Freddi_Fonet
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309090 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines
Merged revisions 309033-309034 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines
A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error.
Detect whether Flex will compile without the workaround; if so, suppress our workaround code.
........
r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines
Clarify meaning, removing double negative (stupid!)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309036 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r308991 | tilghman | 2011-02-28 03:33:22 -0600 (Mon, 28 Feb 2011) | 14 lines
Merged revisions 308990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) | 7 lines
Statements updating zero rows may return SQL_NO_DATA. This is fine; it's handled.
(closes issue #18815)
Reported by: irroot
Patches:
func_odbc.insert_nodata.patch uploaded by irroot (license 52)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308992 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines
Fix Deadlock with attended transfer of SIP call
Call path
sip_set_rtp_peer (locks chan then pvt)
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper (locks p->owner)
But by the time p->owner lock was attempted, seems as though chan and p->owner were different.
So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods.
(closes issue #18837)
Reported by: alecdavis
Patches:
bug18837-trunk.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, Irontec, ZX81, cmaj
Review: [https://reviewboard.asterisk.org/r/1126/]
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308946 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines
Invalid read in ast_channel_set_caller_event().
Valgrind reported that ast_channel_set_caller_event() was reading data
from a freed buffer when using the pre_set structure.
Rearange things to pre-calculate the name and number pointer before
updating the caller party structure to see if the name or number was
changed.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308904 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines
Merged revisions 308814 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines
Merged revisions 308813 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines
Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection every
time someone connects via AMI. It should only be sent to the user who
just connected.
(closes issue #18168)
Reported by: FeyFre
Patches:
bug0018168.patch uploaded by FeyFre (license 1142)
Tested by: FeyFre, twilson
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308816 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r308723 | mnicholson | 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) | 16 lines
Merged revisions 308722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r308722 | mnicholson | 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) | 9 lines
Merged revisions 308721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) | 2 lines
silence gcc 4.2 compiler warning
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308724 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines
Merged revisions 308678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines
Use remotesecret to authenticate with a remote party
The remotesecret option was only being used for outbound registration
and not for placing calls. This patch uses remotesecret on outbound
calls if it is set, otherwise secret is still used.
Review: https://reviewboard.asterisk.org/r/1107/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308680 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308624 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines
sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails.
(closes issue #18874)
Reported by: cmaj
Patches:
patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830)
JIRA SWP-3172
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308623 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Guessed the log levels based on info that level 3
is the soft roof. Can we create a page / document
to define the levels?
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308527 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r308416 | mnicholson | 2011-02-21 09:02:20 -0600 (Mon, 21 Feb 2011) | 19 lines
Merged revisions 308414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines
Merged revisions 308413 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines
Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway.
AST-2011-002
FAX-281
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308417 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
enable reporting of the request URI / URL in debugging
change funny debug note to a serious note.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308372 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
The callback handle_statechange (pbx.c) fails to release its data
pointer, leaking memory in the process.
Reported by: tzafrir
Patches:
18735_pbx_free_callback.diff uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/1110/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308371 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Modern browsers are checking for the MIME Type of pages
and in some cases will not load a file if the type is
wrong.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308331 f38db490-d61c-443f-a65b-d21fe96a405b
|