Age | Commit message (Collapse) | Author |
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(issue ASTERISK-19298)
Reported by: Dmitry Melekhov
Patches:
change_gk_on_reload-1.patch (License #5415)
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Review: https://reviewboard.asterisk.org/r/1770/
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Review: https://reviewboard.asterisk.org/r/1753/
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This change permits each verbose destination (consoles, logger) to have its
own concept of what the verbosity level is. The big feature here is that
the logger will now be able to capture a particular verbosity level without
condemning each console to need to suffer that level of verbosity.
Additionally, a stray 'core set verbose' will no longer change what will go
to the log.
Review: https://reviewboard.asterisk.org/r/1599/
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(Closes issue ASTERISK-19298)
Reported by: robinfood
Patches:
issue19298.patch uploaded by may213 (License #5415)
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Review: https://reviewboard.asterisk.org/r/1733/
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Patch by: Clod Patry
Review: https://reviewboard.asterisk.org/r/1651
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Review: https://reviewboard.asterisk.org/r/1707/
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sounds).
(Closes issue ASTERISK-19233)
Reported by: Matt Behrens
Patches:
chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
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Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.
Review: https://reviewboard.asterisk.org/r/1661/
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r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines
Ensure that two prerequisites are properly installed on Debian-style distributions.
* Don't specify a specific version of libgmime; newer versions are available
now and acceptable.
* Install libsrtp so that res_srtp can be built.
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r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines
Correct some 'set-but-not-used' variable warnings.
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There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
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r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines
Merged revisions 346762 via svnmerge from
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r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines
process null frame pointer returned by ast_rtp_instance_read correctly
(closes issue ASTERISK-16697)
Reported by: under
Patches:
segfault.diff (License #5871) patch uploaded by under
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verbosity level.
Review: https://reviewboard.asterisk.org/r/1599
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Some PBXes require this for call status checking
(closes issue ASTERISK-18748)
Reported by: Fabrizio Lazzaretti
Patches:
ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
Tested by: Fabrizio Lazzaretti
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(these memleaks stop development of gk codes, now i can continue)
Fix printHandler 'Unbalanced Structure' issues with locking printHandler
data for single thread.
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r341313 | may | 2011-10-19 03:33:49 +0400 (Wed, 19 Oct 2011) | 10 lines
Merged revisions 341312 via svnmerge from
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r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3 lines
fix issue on channel numbering (calls could have same channel number
on heavy loaded system)
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r339245 | may | 2011-10-04 15:49:49 +0400 (Tue, 04 Oct 2011) | 9 lines
Merged revisions 339244 via svnmerge from
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r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2 lines
fix forget declaration in previous change
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r339089 | may | 2011-10-03 22:52:55 +0400 (Mon, 03 Oct 2011) | 10 lines
Merged revisions 339087 via svnmerge from
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r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4 lines
destroy memheap mutex properly before memheap deleted
(fix memory leak occured after r304950 changes with DEBUG_THREAD compile option)
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r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) | 6 lines
Remove the channel function OOH323() and place its options into
CHANNEL()
channel drivers should not have there own dialplan functions.
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r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
Merged revisions 335064 via svnmerge from
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r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
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r333961 | may | 2011-08-31 01:21:53 +0400 (Wed, 31 Aug 2011) | 11 lines
Merged revisions 333947 via svnmerge from
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r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines
cleanups in ACF/ARJ GK replies processing
fixed long (24 sec) pause if acf/arj proccessed
before ast_cond_wait called to wait this
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r333962 | may | 2011-08-31 01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines
security fix. really drop call if signalling addr is not same as socket
addr
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r333786 | rmudgett | 2011-08-29 16:12:29 -0500 (Mon, 29 Aug 2011) | 13 lines
Merged revisions 333784-333785 via svnmerge from
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r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011) | 2 lines
Fix deadlock potential of chan_mobile.c:mbl_ast_hangup().
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r333785 | rmudgett | 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line
Add some do not hold locks notes to channel.h
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r331147 | may | 2011-08-09 20:16:55 +0400 (Tue, 09 Aug 2011) | 11 lines
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r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4 lines
move ast_cond_signal for admitted call after all data filled/freed
clear all log channels by pointed number not only first
free allocated callToken in ooh323_answer
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r331200 | may | 2011-08-09 20:36:39 +0400 (Tue, 09 Aug 2011) | 9 lines
Setup IP proto version for call in GK mode
Added additional check for IP semantics before parse destination
by ast_parse_args due to it can parse numeric as IP.
(closes issue ASTERISK-18218)
Reported by: slesru
Patch: ASTERISK-18218.patch
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r330899 | may | 2011-08-05 11:38:28 +0400 (Fri, 05 Aug 2011) | 11 lines
Merged revisions 330827 via svnmerge from
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r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4 lines
change gk client behaivour on rrq/grq failures to setup timers
and next tries after timeout instead of complete failure in the ooh323
stack
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r328428 | may | 2011-07-15 23:31:09 +0400 (Fri, 15 Jul 2011) | 13 lines
Merged revisions 328427 via svnmerge from
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r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 lines
small gk processing fixes:
- decrease for 1 second registration ttl for very low expirations (some
providers expire few earlier than TTL)
- delete rrq and registration expire timers on URQ received as we make
new registration.
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r328429 | may | 2011-07-15 23:35:50 +0400 (Fri, 15 Jul 2011) | 2 lines
delete unproperly changed svn props
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
Merged revisions 328209 via svnmerge from
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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Add full t.38 handshaking for OOH323 that are required for newest T.38
gateway codes.
Add fax detection (cng tone, t38) and dialplan redirection to fax ext on
fax event detected.
Add OOH323() function to set/get t38support and faxdetect parameters.
(closes issue ASTERISK-17754)
Reported by: irroot
Patches:
ooh323_faxdetect.patch uploaded by irroot (license 52)
issue19183-final.patch uploaded by may213 (license 454)
Tested by: may213, irroot
Review: https://reviewboard.asterisk.org/r/1174/
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r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines
Fix double alerting, add forced alerting before answer
Fix double alerting (it wasn't fixed here by issue #18542)
Add forced alerting before connect (if it wasn't before)
Try to send all packets from outgoing queue rather than one only
Call goes into clearing state when disconnect command is received
(closes issue #19361)
Reported by: vmikhelson
Patches:
issue19361-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
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r320716 | twilson | 2011-05-24 17:49:10 -0700 (Tue, 24 May 2011) | 4 lines
Cast data as char * before using S_OR
This is required for compiling successfully under dev mode
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r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) | 11 lines
Fix a crash in the MySQL() application.
This code was not handling channel datastores safely. The channel
must be locked.
(closes issue #17964)
Reported by: wuwu
Patches:
issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license 71)
Tested by: wuwu
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r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May 2011) | 10 lines
Don't duplicate our data on the stack and just use the MYSQL_ROW directly.
With large result sets we were blowing out the stack.
(closes issue #19090)
Reported by: mickecarlsson
Patches:
issue19090_trunk_svn.patch uploaded by seanbright (license 71)
Tested by: mickecarlsson
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r316429 | tilghman | 2011-05-03 19:12:25 -0500 (Tue, 03 May 2011) | 7 lines
Escape column names in case they contain illegal characters ('-') or reserved words.
(closes issue #19063)
Reported by: festr
Patches:
patch uploaded by festr (license 443)
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IPv6 support for ooh323,
bindaddr, peers and users ip can be IPv4 or IPv6 addr
correction for multi-homed mode (0.0.0.0 or :: bindaddr)
can work in dual 6/4 mode with :: bindaddr
gatekeeper mode isn't supported in v6 mode while
(issue #18278)
Reported by: may213
Patches:
ipv6-ooh323.patch uploaded by may213 (license 454)
Review: https://reviewboard.asterisk.org/r/1004/
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r313142 | may | 2011-04-10 00:56:17 +0400 (Sun, 10 Apr 2011) | 3 lines
fix trivial bug in ooh323_indicate on AST_CONTROL_SRC...
check p->rtp is not null
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r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines
Reload must react correctly against a possibly changed table, so dropping the conditional reload flag.
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r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines
Merged revisions 312287 via svnmerge from
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r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines
Merged revisions 312285 via svnmerge from
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r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines
Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code.
(issue #18969)
Reported by: oej
Patches:
20110315__issue18969__14.diff.txt uploaded by tilghman (license 14)
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r311687 | may | 2011-03-28 01:47:13 +0400 (Mon, 28 Mar 2011) | 2 lines
correct return values in ooh323_indicate for AST_CONTROL_T38_PARAMETERS
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r310834 | tilghman | 2011-03-14 20:48:25 -0500 (Mon, 14 Mar 2011) | 2 lines
Fix branch compile.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
(closes issue #18693)
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 lines
added g729onlyA option for announce only AnnexA g.729 codec in
h.323 capabilities. Option can be global or per user/peer.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb 2011) | 2 lines
Fix FreeBSD builds.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 lines
ifdef __linux__ keepalive variables also
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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don't have these options on sockets.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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change malloc to ast_calloc again
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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small fixes.
Interpret remote side H.225 version.
Corrections for H.323v2 endpoints:
don't start TCS and MSD before connect,
don't start TCS and MSD by accepting H.245 connection,
start TCS and MSD by StartH245 facility message.
Other fixes:
fix non zeroended remoteDisplayName issue, small fixes in call clearing
by closing H.245 connection, tcp keepalive introduced on TCP
connections (now is hardcoded, will be configurable in the future),
don't force H.245tunneling if FastStart is active, don't send Alerting
singal more than once per call.
(closes issue #18542)
Reported by: vmikhelson
Patches:
issue18542-final-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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