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r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines
Merged revisions 199138 via svnmerge from
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r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines
Additional updates to AST-2009-001
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r199091 | eliel | 2009-06-04 12:29:50 -0400 (Thu, 04 Jun 2009) | 11 lines
Move static docs to the new AstXML form.
Move SMDI_MSG and SMDI_MSG_RETRIEVE functions statis documentation
to XML.
(issue #15245)
Reported by: eliel
Patches:
res_smdi_static_conversion.txt uploaded by lmadsen (license 10)
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r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines
Merged revisions 199022 via svnmerge from
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r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines
Safely handle AMI connections/reload requests that occur during startup.
During asterisk startup, a lock on the list of modules is obtained by the
primary thread while each module is initialized. Issue 13778 pointed out a
problem with this approach, however. Because the AMI is loaded before other
modules, it is possible for a module reload to be issued by a connected client
(via Action: Command), causing a deadlock.
The resolution for 13778 was to move initialization of the manager to happen
after the other modules had already been lodaded. While this fixed this
particular issue, it caused a problem for users (like FreePBX) who call AMI
scripts via an #exec in a configuration file (See issue 15189).
The solution I have come up with is to defer any reload requests that come in
until after the server is fully booted. When a call comes in to
ast_module_reload (from wherever) before we are fully booted, the request is
added to a queue of pending requests. Once we are done booting up, we then
execute these deferred requests in turn.
Note that I have tried to make this a bit more intelligent in that it will not
queue up more than 1 request for the same module to be reloaded, and if a
general reload request comes in ('module reload') the queue is flushed and we
only issue a single deferred reload for the entire system.
As for how this will impact existing installations - Before 13778, a reload
issued before module initialization was completed would result in a deadlock.
After 13778, you simply couldn't connect to the manager during startup (which
causes problems with #exec-that-calls-AMI configuration files). I believe this
is a good general purpose solution that won't negatively impact existing
installations.
(closes issue #15189)
(closes issue #13778)
Reported by: p_lindheimer
Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer, seanbright
Review: https://reviewboard.asterisk.org/r/272/
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r198958 | seanbright | 2009-06-03 16:49:11 -0400 (Wed, 03 Jun 2009) | 17 lines
Blocked revisions 198957 via svnmerge
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r198957 | seanbright | 2009-06-03 16:39:10 -0400 (Wed, 03 Jun 2009) | 11 lines
Fix a possible crash in pbx_spool.
We were trying to reference members of a struct that had previously been freed.
This patch makes sure that we free the struct after it has been removed from
the spooler queue.
(closes issue #15072)
Reported by: garlew
Patches:
spool.diff uploaded by garlew (license 376)
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r198954 | dvossel | 2009-06-03 15:30:10 -0500 (Wed, 03 Jun 2009) | 3 lines
ast_call_forward() todo notes and originate flag copy.
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r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
Generic call forward api, ast_call_forward()
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
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r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines
fixes issue with channels not going down after transfer
Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop.
(closes issue #15216)
Reported by: oxymoron
Tested by: dvossel
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r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
Correct documentation for the register line, specifically where the domain should be specified.
(closes issue #14367)
Reported by: Nick_Lewis
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r198762 | file | 2009-06-02 10:12:59 -0300 (Tue, 02 Jun 2009) | 5 lines
Fix a bug where we were passing in address information that should remain unmodified to a function that may modify it.
(closes issue #15243)
Reported by: pj
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r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009) | 2 lines
Tell the IAX2 parser about more control frame types.
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r198727 | mmichelson | 2009-06-01 15:57:31 -0500 (Mon, 01 Jun 2009) | 13 lines
Add the ability to execute connected line interception macros.
When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
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r198725 | tilghman | 2009-06-01 15:33:50 -0500 (Mon, 01 Jun 2009) | 8 lines
Add INCrement and DECrement functions
(closes issue #15025)
Reported by: greenfieldtech
Patches:
func_math.c.patch_v4 uploaded by greenfieldtech (license 369)
slightly modified by me
Tested by: greenfieldtech, lmadsen
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r198670 | russell | 2009-06-01 15:17:50 -0500 (Mon, 01 Jun 2009) | 2 lines
Minor whitespace fix.
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r198661 | eliel | 2009-06-01 15:37:30 -0400 (Mon, 01 Jun 2009) | 6 lines
Moved more static documentation to the new AstXML form.
Moved more static docs to XML (pplications and manager actions):
Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor.
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r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 Jun 2009) | 2 lines
Add information for new meetme realtime fields
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r198597 | eliel | 2009-06-01 13:53:38 -0400 (Mon, 01 Jun 2009) | 3 lines
Do not add say.o in a separate line.
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r198565 | eliel | 2009-06-01 12:09:42 -0400 (Mon, 01 Jun 2009) | 2 lines
Move JabberSend manager action from static docs to the AstXML form.
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r198561 | eliel | 2009-06-01 11:38:48 -0400 (Mon, 01 Jun 2009) | 2 lines
Move static documentation of E|Dead|AGI() application and manager action to XML.
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r198529 | mmichelson | 2009-06-01 09:42:57 -0500 (Mon, 01 Jun 2009) | 5 lines
Remove extra lock from local_indicate in connected line case.
Oh, and this fixes a deadlock I was seeing.
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r198530 | mmichelson | 2009-06-01 09:45:43 -0500 (Mon, 01 Jun 2009) | 3 lines
Remove extra lock from app_queue.
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r198500 | mmichelson | 2009-06-01 09:02:05 -0500 (Mon, 01 Jun 2009) | 6 lines
Remove documentation for the 'exten' argument to the AGENT function.
Since AgentCallbackLogin has been removed, this should not be documented
any more.
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r198498 | file | 2009-06-01 10:31:27 -0300 (Mon, 01 Jun 2009) | 5 lines
Fix a bug where the Event and Content-Type headers were added twice to outgoing SIP NOTIFY messages.
(closes issue #15239)
Reported by: pj
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r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009) | 2 lines
Fix documentation for FIELDQTY.
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r198442 | eliel | 2009-05-30 22:09:06 -0400 (Sat, 30 May 2009) | 3 lines
Filter the say.o object, it is being added later.
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r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) | 11 lines
Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded.
if dahdi_test_timer() fails, timing_funcs_handle remains NULL causing a crash
when calling ast_unregister_timing_interface() with a NULL pointer.
(closes issue #15234)
Reported by: eliel
Patches:
timing_dahdi1.diff uploaded by eliel (license 64)
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r198438 | russell | 2009-05-30 20:40:02 -0500 (Sat, 30 May 2009) | 2 lines
Constification and remove some unused code.
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r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) | 12 lines
Merged revisions 198311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines
Fix a crash that occurred when MWI SMDI messages expired.
(closes issue #14561)
Reported by: cmoss28
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r198434 | russell | 2009-05-30 20:19:30 -0500 (Sat, 30 May 2009) | 2 lines
Constify the ast_frame arg to ast_queue_frame().
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r198375 | seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 lines
Properly terminate the receive buffer before sending to iksemel.
aji_io_recv takes the maximum number of bytes to read (instead of the total
buffer size), so we have to subtract 1 from our buffer size. Without this, when
we receive packets that are larger than our buffer, iksemel will choke and
things get wonky.
(closes issue #15232)
Reported by: lp0
Patches:
05302009_res_jabber.c.patch uploaded by seanbright (license 71)
Tested by: seanbright, lp0
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r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May 2009) | 19 lines
Merged revisions 198370 via svnmerge from
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r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines
Properly terminate AMI JabberSend response messages.
The response message (either Error or Success) needs an extra trailing \r\n
after the fields to inform the client that the message is complete.
(closes issue #14876)
Reported by: srt
Patches:
05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71)
asterisk_14876.patch uploaded by srt (license 378)
trunk-14876-2.diff uploaded by phsultan (license 73)
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r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines
Merged revisions 198251 via svnmerge from
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r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines
Treat an empty FORWARD_CONTEXT the same way we treat a missing one.
(closes issue #15056)
Reported by: p_lindheimer
Patches:
05292009_bug15056.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer
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r198248 | file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines
When removing all packets from a dialog we also need to free the data if present.
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r198217 | eliel | 2009-05-29 21:04:57 -0400 (Fri, 29 May 2009) | 10 lines
Remove not used code in the Agent channel.
This code was there because of the AgentCallbackLogin() application.
->loginchan[] member was only used by AgentCallbackLogin().
Agent where dumped to astdb if they where logged in using AgentCallbacklogin()
so they are not being dumper anymore.
Review: https://reviewboard.asterisk.org/r/267/
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r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29 May 2009) | 2 lines
Suggesting that only a single timing module be loaded is no longer necessary.
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r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009) | 2 lines
Improve handling of trying to ACK too many timer expirations.
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r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009) | 38 lines
Resolve issues with choppy sound when using res_timing_pthread.
The situation that caused this problem was when continuous mode was being
turned on and off while a rate was set for a timing interface. A very easy
way to replicate this bug was to do a Playback() from behind a Local channel.
In this scenario, a rate gets set on the channel for doing file playback.
At the same time, continuous mode gets turned on and off about every 20 ms
as frames get queued on to the PBX side channel from the other side of the
Local channel.
Essentially, this module treated continuous mode and a set rate as mutually
exclusive states for the timer to be in. When I dug deep enough, I observed
the following pattern:
1) Set timer to tick every 20 ms.
2) Wait almost 20 ms ...
3) Continuous mode gets turned on for a queued up frame
4) Continuous mode gets turned off
5) The timer goes back to its tick per 20 ms. state but starts counting
at 0 ms.
6) Goto step 2.
Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick,
but not most of the time. This is what produced the choppy sound (or sometimes
no sound at all).
Now, the module treats continuous mode and a set rate as completely independent
timer modes. They can be enabled and disabled independently of each other and
things work as expected.
(closes issue #14412)
Reported by: dome
Patches:
issue14412.diff.txt uploaded by russell (license 2)
issue14412-1.6.1.0.diff.txt uploaded by russell (license 2)
Tested by: DennisD, russell
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r198088 | jpeeler | 2009-05-29 14:19:51 -0500 (Fri, 29 May 2009) | 9 lines
New signaling module to handle analog operations in chan_dahdi
This branch splits all the analog signaling logic out of chan_dahdi.c into
sig_analog.c. Functionality in theory should not change at all. As noted
in the code, there is still some unused code remaining that will be cleaned
up in a later commit.
Review: https://reviewboard.asterisk.org/r/253/
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r198139 | eliel | 2009-05-29 15:46:07 -0400 (Fri, 29 May 2009) | 15 lines
Simplify the Makefile and avoid needing to specify each object file.
Instead of specifying every object file, use make's magic to generate
it.
This will generate less conflicts in team branches when a new file is
added in trunk.
(closes issue #15226)
Reported by: eliel
Patches:
makefile uploaded by eliel (license 64)
Review: http://reviewboard.asterisk.org/r/269/
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r198083 | eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines
Apply anti-spam obfuscation to an email address.
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r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines
Merged revisions 198068 via svnmerge from
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r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines
Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition.
This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
(closes issue #12946)
Reported by: meral
Patches:
null-cdr2.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, dbrooks
(closes issue #15122)
Reported by: sum
Tested by: sum
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r198064 | file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines
Fix a memory leak of the write buffer when writing a file.
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r197959 | mmichelson | 2009-05-29 10:48:04 -0500 (Fri, 29 May 2009) | 16 lines
A few fixes to SIP with regards to connected line updates during transfers.
* Set the invitestate to INV_CALLING when we send a connected line reinvite.
This prevents us from potentially rapid-firing reinvites to a single peer.
* Use the astdb to store a peer's allowed methods. This prevents us from sending
an UPDATE during the interval between startup and the peer's first registration
if the peer does not support the UPDATE method.
* Handle Polycom's method of indicating allowed methods in REGISTER. Instead of
using an Allow header, they place the allowed methods in a methods= parameter
in the Contact header.
ABE-1873
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r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May 2009) | 15 lines
Merged revisions 197998 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May 2009) | 8 lines
Fix 'make config' target for Slackware.
There was a missing semi-colon after the echo statement in the Makefile that was
causing problems for some users. Fix suggested by reporter.
(closes issue #15225)
Reported by: pdavis
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r197996 | file | 2009-05-29 14:51:06 -0300 (Fri, 29 May 2009) | 2 lines
Fix a bug where the default setting did not perform a remote bridge when it should have.
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r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009) | 2 lines
Trim trailing whitespace so that I can work on this bug without it bothering me. :-)
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r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009) | 8 lines
Update documentation in MixMonitor.
Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize away the Local
channel when using this option.
(closes issue #14829)
Reported by: licedey
Tested by: mmichelson, licedey, lmadsen
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r197861 | seanbright | 2009-05-28 18:42:27 -0400 (Thu, 28 May 2009) | 2 lines
Update references to downloads.digium.com to its new URL.
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r197824 | seanbright | 2009-05-28 17:50:27 -0400 (Thu, 28 May 2009) | 2 lines
Update references to bugs.digium.com and reviewboard.digium.com to the new URLs.
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r197738 | twilson | 2009-05-28 14:57:18 -0500 (Thu, 28 May 2009) | 19 lines
Add Calendaring support for Asterisk
This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS
Exchange calendars. Exchange support has only been tested on Exchange Server 2k3
and does not support forms-based authentication at this time (patches *very*
welcome). Exchange support is also currently missing the ability to return a
list of a meting's attendees (again, patches are very, very welcome).
Features include:
Querying a calendar for events over a specific time range
Checking a calendar's busy status via the dialplan
Writing calendar events via the dialplan (CalDAV and Exchange only)
Handling calendar event notifications through the dialplan
(closes issue #14771)
Tested by: lmadsen, twilson, Shivaprakash
Review: https://reviewboard.asterisk.org/r/58
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r197775 | kpfleming | 2009-05-28 15:36:49 -0500 (Thu, 28 May 2009) | 10 lines
Ensure that accidental calls to ast_string_field_free_memory() on embedded stringfield pools are safe.
It is possible for a stringfield manager structure (and pool) structure to be allocated
as part of a larger structure allocation (using ast_calloc_with_strinfields()); when
this is done, the stringfield pool cannot be separately freed, but users of the tructure
may not be aware (and shouldn't have to be aware) of whether the pool was embedded.
This patch modifies the behavior so that they can always call ast_string_field_free_memory()
and the function will do the right thing for both embedded and non-embedded situations.
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r197740 | mmichelson | 2009-05-28 15:17:24 -0500 (Thu, 28 May 2009) | 7 lines
Treat 405 responses the same way we would a 501.
This makes sure that we mark a method as being unallowed if we
receive a 405 response so that we don't continue to try to
send that same type of message.
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