When retrieving the parking lots from a MySQL database table, the current order
is "filename, cat_metric desc, var_metric asc, category". If there are multiple
parking lots with the same cat_metric but different categories, everything is
being sorted on cat_metric first resulting in errors when loading the parking
lots.
This patch fixes the problem by sorting on the category field first, then the
cat_metric field.
(closes issue ASTERISK-21035)
Reported by: Alex Epshteyn
Patches:
asterisk-21035-orderby.diff Michael L. Young (license 5026)
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In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed
to include the Username field. Somehow, one of the events was missed. This
patch corrects that - the Username field should be included in all AMI Registry
events involving SIP registrations.
(issue ASTERISK-17888)
(closes issue ASTERISK-21201)
Reported by: Dmitriy Serov
patches:
chan_sip.c.diff uploaded by Dmitriy Serov (license 6479)
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Added an option "discard_remote_hold_retrieval" (default "no") that if set does
not trigger the music on hold event. This essentially stops telling the peer
to start music on hold.
(issue ABE-2899)
Reported by: Denis Alberto Martinez
Review: https://reviewboard.asterisk.org/r/2336/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this change, certain conditions for sending the message would
result in an address of '(null)' being used in the via header of the
SIP message because a NULl value of pvt->ourip was used when initially
generating the via header. This is fixed by adding a call to build_via
when the address is set before sending the message.
(closes issue ASTERISK-21148)
Reported by: Zhi Cheng
Patches:
700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475)
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The cache dump mechanism allows the developer to retreive multiple
items of a given type (or of all types) from the cache residing in a
stasis caching topic in addition to the existing single-item cache
retreival mechanism. This also adds to the caching unit tests to
ensure that the new cache dump mechanism is functioning properly.
Review: https://reviewboard.asterisk.org/r/2367/
(issue ASTERISK-21097)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For the initial use of this bus, I took some work kmoore did creating
channel snapshots. So rather than create AMI events directly in the
channel code, this patch generates Stasis events, which manager.c uses
to then publish the AMI event.
This message bus provides a generic publish/subscribe mechanism within
Asterisk. This message bus is:
- Loosely coupled; new message types can be added in seperate modules.
- Easy to use; publishing and subscribing are straightforward
operations.
In addition to basic publish/subscribe, the patch also provides
mechanisms for message forwarding, and for message caching.
(issue ASTERISK-20887)
(closes issue ASTERISK-20959)
Review: https://reviewboard.asterisk.org/r/2339/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Originally, way back in r201583, we added the alternate RTP address so
that the RTP engine would expect to receive audio from a new source
when a glare re-INVITE occurred. In r382589, we remove the alternate
RTP source, as the 'secret' probation mode allows for switching to a new
RTP source when a previous source stops sending RTP. At the time, it
seemed appropriate to set the RTP source based on the information in the
glared re-INVITE.
Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs
with no SDP - such as in a connected line update that glances - we'll set
the RTP source to an invalid address. In subsequent re-INVITE requests from
this Asterisk instance, we'll then send an invalid media address, which will
result in the remote side sending a 488. Whoops.
There isn't any need to reset the RTP source - if we're using strictrtp, we'll
simply synchronize to a new source when we stop getting packets from the old
one. If we aren't using strictrtp, then again there shouldn't be a problem.
Note that the Asterisk Test Suite's connectedline test caught this error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot
that Urgent also "counts" as new messages. This fixed the problem when any of
the three folders was specified and the combine option was used.
It missed the case where the folder isn't specified and we build a snapshot of
all folders. This patch corrects that.
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Often, Asterisk may realize that a change in the source of an RTP stream is
about to occur and ask that the RTP engine reset it's lock on the current RTP
source. In certain scenarios, it may take awhile for the new remote system to
send RTP packets, while the old remote system may continue providing RTP during
that time period. This causes Asterisk to re-lock onto the old source, thereby
rejecting the new source when the old source stops sending RTP and the new
source begins.
This patch prevents that by having a constant secondary, 'secret' probation
mode enabled when an RTP source has been chosen. RTP packets from other sources
are always considered, but never chosen unless the current RTP source stops
sending RTP.
Review: https://reviewboard.asterisk.org/r/2364
(closes issue AST-1124)
Reported by: John Bigelow
Tested by: John Bigelow
(closes issue AST-1125)
Reported by: John Bigelow
Tested by: John Bigelow
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If ast_threadpool_set_size with a size equal to the current size, a
reference to a set_size_data structure would be leaked.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ownership of the listener reference is not transferred because the
listener is reffed when placed into the taskprocessor. Ensure that the
listener is dereffed properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.
A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.
Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/
(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
oolong-path-support-trunk in team branch by oej (License 5267)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fix checks to make sure that if a confbridge record start command is issued
from the CLI it will always use the file name given on the CLI even if it
changes between start/stop records for a conference. Previously it had been
reusing the same file between start/stops even if a new filename was given.
(issue AST-1088)
Reported by: John Bigelow
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The original report had to do with a realtime peer behind NAT being pruned and
the peer's private address being used instead of its external address. Upon
debugging, it was discovered that this was being caused by the addition of
the auto_force_rport and auto_comedia settings.
This patch does the following:
* Adds a missing note to the CHANGES file indicating that the default global nat
setting is auto_force_rport
* Constify the 'req' parameter for check_via()
* Add calls to check_via() in a couple of places in order for the auto_*
settings to do their job in attempting to determine if NAT is involved
* Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_*
settings are in use where it was needed
* Moves the copying of peer flags up in build_peer() to before they are used;
this fixes the realtime prune issue
* Update the contrib/realtime schemas to allow the nat column to handle the
different nat setting combinations we have
This patch received a review and "Ship It!" on the issue itself.
(closes issue ASTERISK-20904)
Reported by: JoshE
Tested by: JoshE, Michael L. Young
Patches:
asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026)
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If the end result of the ICE negotiation resulted in the path for media
changing it was possible for the strictrtp code to discard the RTP packets.
This change causes strictrtp to enter learning mode once again when the
ICE negotiation has completed successfully.
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A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it
already holds the iax2 private lock and improperly fails to obtain the channel
lock before calling ast_set_callerid. By not safely obtaining the channel lock,
a locking inversion can take place, causing a deadlock.
This patch solves this by calling the required deadlock avoidance functions
that obtain the channel lock before setting the caller ID.
Thanks to Pavel for fixing my syntax errors and testing this patch out.
(closes issue ASTERISK-21128)
Reported by: Pavel Troller
Tested by: Pavel Troller
patches:
ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302)
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For some channel drivers, specifically those that have a varying rate in the
number of audio samples, the audio quality for a MeetMe conference can be
exceedingly poor. This is due to a unilateral application of the DENOISE
function in func_speex to channels joining the conference.
The denoiser function in the speex library is initialized with the number of
audio samples in each sample that will be provided to it. If the number of
audio samples changes, the denoiser has to be thrown away and re-initialized.
While this could be worked around by removing func_speex, that doesn't help
if you actually use the denoiser with other channels on the system.
This patches does the following:
* Checks for the presence of func_speex as opposed to codec_speex when
determining if the DENOISE function is present (which is where the function
is actually implemented)
* Adds an option to MeetMe 'n' that causes the denoiser to not be applied
to a channel when it joins. This keeps the current behavior the default, but
let's users disable the denoiser if it causes problems on their system.
Review: https://reviewboard.asterisk.org/r/2358
(closes issue AST-1062)
Reported by: Thomas Arimont
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When ConfBridge was refactored to better handle the concept of marked,
wait_marked, and normal users co-existing in a conference (thereby implementing
a state machine for the conference), the wait_marked users were put into their
own list of conference participants, separate from the active users. This list
is used for wait_marked users when they are waiting in a conference but no
marked user has joined; normal users may have joined at this point however.
There are several AMI/CLI commands that affect conference users that were not
checking the wait_marked users list:
* CLI/AMI commands that mute/unmute a participant. In this case, wait_marked
users have to remain in their particular state and should not be affected -
however, the commands would return "Channel not found" as opposed to the
appropriate error condition.
* CLI/AMI commands that kick a participant. An admin should always be able to
kick a participant out of the conference.
This patch fixes both sets of commands, and cleans up the CLI commands slightly
by allowing them to complete a participant name (this was supposed to have been
added, but the function call was commented out and wasn't implemented).
Review: https://reviewboard.asterisk.org/r/2346/
(closes issue AST-1114)
Reported by: John Bigelow
Tested by: John Bigelow
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ConfBridge and Page require that there always be a default bridge and user
profile available. While properties of the default profiles can be overriden
in the configuration file, removing them can create situations where neither
application can function properly.
This patch ensures that if an administrator removes the profiles from the
confbridge.conf configuration file, the profiles are added upon load.
Documentation clarifying this has been added to the confbridge.conf.sample file.
Review: https://reviewboard.asterisk.org/r/2356/
(closes issue AST-1115)
Reported by: John Bigelow
Tested by: John Bigelow
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There were several problems using variadic argument macros in res_config_mysql.
* Improper use of va_end. Multiple calls to va_end were possible resulting in
an unbalanced matching of va_start/va_end.
* Calls to va_arg after a possible encounter of a SENTINEL value.
This patch corrects those errors.
(closes issue ASTERISK-19451)
Reported by: wdoekes
patches:
ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Basically sets the callerid and callername to the first device talked to for the
purposes of putting the the calls made log on the device. Does not affect the device
displaying who the device is currently talking to.
Also some minor changes to use sub->exten in lieu of l->lastnumberdialed.
(closes issue ASTERISK-21095)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-calllogsoutbound03.diff uploaded by wedhorn (license 5019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
