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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197861 f38db490-d61c-443f-a65b-d21fe96a405b
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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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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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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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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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1240 on load but was on reload.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines
Use the address we already know when reloading a peer with nat=yes.
If we already have an address for a peer, and we are reloading the sip
configuration, try to use that address to contact the peer, instead of
getting it from the Contact.
(closes issue #15194)
Reported by: ibc
Patches:
sip.patch uploaded by eliel (license 64)
Tested by: manwe
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(closes issue #14833)
Reported by: contactmayankjain
Patches:
all_changes.patch uploaded by contactmayankjain (license 740)
slightly modified by me
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
Allow for media to arrive from an alternate source when responding to a reinvite with 491.
When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.
As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
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in the logger.
Since the message structure is allocated using ast_calloc_with_stringfields we do not need to
free the memory used for the stringfields as it will get freed when the message structure is.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
Add flags to chanspy audiohook so that audio stays in sync.
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
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(closes issue #15074)
Reported by: pj
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Let's try that again, this time removing trailing whitespace and not leading
whitespace. I can't believe no one noticed.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines
Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting.
The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated
(or it passes through unauthenticated) the proper nat flag is set.
(closes issue #13823)
Reported by: dimas
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Added AstVoicemailContext
(closes issue #15155)
Reported by: scramatte
Tested by: suretec
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Added new ObjectClass AsteriskVoiceMail, and AstAccountCallLimit attribute
and cleaned up formatting and tested with OpenLDAP
(closes issue #15155)
Reported by: scramatte
Patches:
asterisk.schema uploaded by scramatte (license 796)
Tested by: suretec
Review: [full review board URL with trailing slash]
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Previously they always set hangupcause 0, which is generally wrong. With this
change, we're setting some generic hangup causes. For 5xx errors, which indicate
some sort of problem with the remote server, we're now setting CONGESTION.
EDVX002
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r197264 | seanbright | 2009-05-27 16:12:06 -0400 (Wed, 27 May 2009) | 6 lines
Since we use bashisms in build_tools/mkpkgconfig, we should call on bash
explicitly when running from the Makefile.
(closes issue #15209)
Reported by: seandarcy
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Since we use bashisms in build_tools/mkpkgconfig, we should call on bash
explicitly when running from the Makefile, otherwise we get errors during a
'make install.'
(closes issue #15209)
Reported by: seandarcy
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197260 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines
Use a different determinator on whether to print the delimiter, since leading fields may be blank.
(closes issue #15208)
Reported by: ramonpeek
Patch by me, though inspired in part by a patch from ramonpeek
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r197124 | jpeeler | 2009-05-27 11:49:38 -0500 (Wed, 27 May 2009) | 12 lines
Fix broken attended transfers
The bridge was terminating immediately after the attended transfer was
completed. The problem was because upon reentering ast_channel_bridge
nexteventts was checked to see if it was set and if so could possibly
return AST_BRIDGE_COMPLETE.
(closes issue #15183)
Reported by: andrebarbosa
Tested by: andrebarbosa, tootai, loloski
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the sample configuration files.
(closes issue #15207)
Reported by: seandarcy
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r197024 | seanbright | 2009-05-27 09:54:35 -0400 (Wed, 27 May 2009) | 17 lines
Fix handling of the 'state_interface' option of the 'queue add member' CLI
command.
This change relates to r184980, which was a backport of the state interface
changes to app_queue from trunk. trunk and all of the 1.6.x branches are not
affected.
'queue add member' allows for specifying an interface to use for device state
when adding a queue member via CLI, but the validation code was not properly
updated to reflect this optional argument.
(closes issue #15198)
Reported by: loloski
Patches:
05272009_app_queue.diff uploaded by seanbright (license 71)
Tested by: loloski
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or inaccessible configuration file.
Before this change, when chan_alsa failed to load due to a missing or
inaccessible configuration file, no message would be displayed. With this
change, when chan_alsa fails to load due to a missing or inaccessible
configuration file, a message will be displayed.
(closes issue #14760)
Reported by: Nick_Lewis
Patches:
chan_alsa.c-confload.patch uploaded by Nick (license 657)
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(closes issue #15200)
Reported by: ajohnson
Patches:
05262009_xmldoc.patch uploaded by seanbright (license 71)
Tested by: ajohnson
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(closes issue #14988)
Reported by: tzafrir
Patches:
configure.ac.diff uploaded by homesick (license 91)
new_ast_check_osptk.m4 uploaded by homesick (license 91)
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There is inconsistency in how we handle manager responses that are lists of
items and, unfortunately, third parties have come to rely on ActionID being on
every event within those lists instead of just keeping track of the ActionID for
the current response. This change makes CoreShowChannels include the ActionID
with each CoreShowChannel event generated as a result of it being called.
(closes issue #15001)
Reported by: sum
Patches:
patchactionid2.patch uploaded by sum (license 766)
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The CoreStartupTime and CoreReloadTime name/value pairs in the CoreStatus
response message only included the time and not the date. This patch,
inspired by the reporter's patch, adds 2 new fields - CoreStartupDate and
CoreReloadDate - which contain the date portion of these values.
(closes issue #15000)
Reported by: sum
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines
Resolve a file handle leak.
The frames here should have always been freed. However, out of luck, there was
never any memory leaked. However, after file streams became reference counted,
this code would leak the file stream for the file being read.
(closes issue #15181)
Reported by: jkroon
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'channel originate ... application <app>' CLI command.
(And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop,
wanna fight about it!?)
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the peer.
(closes issue #15118)
Reported by: alecdavis
Patches:
chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7 lines
Remove some bash specific stuff from safe_asterisk.
(closes issue #10812)
Reported by: paravoid
Patches:
safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
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Move AGI commands documentation to XML docs:
'set priority'
'set variable'
'stream file'
'control stream file'
'tdd mode'
'verbose'
'wait for digit'
'speech create'
'speech set'
'speech destroy'
'speech load grammar'
'speech unload grammar'
'speech activate grammar'
'speech deactivate grammar'
'speech recognize'
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Move AGI commands ('say datetime', 'send image', 'send text', 'set autohangup',
'set callerid', 'set context', 'set extension') documentation to the AstXML
form.
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substitution is done.
cdr_custom was creating a ast_channel struct directly and passing it into the
core for variable substition. This was fine as long as the format string
contained only calls to the CDR() function. Doing something like ${EPOCH} on
the other hand tried to lock the channel, which would fail and throw an error
because the passed channel hadn't been allocated as an ao2 object. So now we
create the dummy channel with ast_channel_alloc, and everything works as
expected.
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Reported by Atis on #asterisk-dev
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