While looking at ASTERISK-22236, Walter Doekes pointed out that when running
"sip show peers", the setting being displayed can be confusing. The display of
"N" used to mean NAT (i.e. yes). The NAT setting has gone through many
different changes resulting in the display of different characters to try and
convey what the current setting is for 'Forcerport' (A for Auto and Forcerport
is currently on, a for Auto but Forcerport is off, Y for yes, and N for no).
During the initial code review to try and clarify these settings (especially
since "N" no longer meant what it used to mean in prior versions of Asterisk),
Mark Michelson suggested using the full space available to display the settings
which helped to make the settings very clear. That was a great suggestion.
Therefore, this patch does the following:
* The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No.
* A column for the 'Comedia' setting has been added. It too will display the
setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No.
* UPGRADE.txt has been updated to document this change.
(closes issue ASTERISK-22728)
Reported by: Walter Doekes
Tested by: Michael L. Young
Patches:
asterisk-forcerport-display-clarification_v3.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2941
........
Merged revisions 402111 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Surrogate channels would pop up from time to time in dial message handling.
This would cause a WARNING message to appear, indicating that the Surrogate
channel had no CDR. This patch filters out those channels that have the
internal implementation flag set, such that the WARNING message isn't
displayed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Removed another silly use of RAII_VAR(). RAII_VAR() and SCOPED_LOCK()
are not silver bullets that allow you to turn off your brain.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In function ast_rtp_instance_early _bridge_make_compatible the
use of instance 0/1 as arguments doesn't clearly communicate a
direction that the copying of payloads from the source channel
to the destination channel will occur, making it more probable
to have the arguments to ast_rtp_codecs_payloads_copy() put in
the reverse order. This patch renames the arguments with _dst
and _src suffixes and corrects the copy direction.
(closes issue ASTERISK-21464)
Reported by: Kevin Stewart
Review: https://reviewboard.asterisk.org/r/2894/
........
Merged revisions 402000 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Test shows rtpmap:119 being copied per this change, but is not in sip invite
........
Merged revisions 402042 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adapts the behaviour of avpf to only impact the format of outgoing calls. For
inbound calls, both AVP and AVPF calls will be accepted regardless of the value
of avpf in the configuration.
(closes issue ASTERISK-22005)
Reported by: Torrey Searle
Patches:
optional_avpf_trunk.patch uploaded by tsearle (license 5334)
........
Merged revisions 401884 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After a series of upgrades over recent weeks, I've discovered that
test_json.c won't compile in dev mode any more for me.
One of gcc-4.8.2, OS X Mavericks or Xcode 5 has decided to deprecate
tempnam. Which, in general, is a good thing. But for test code that just
needs a temporary file, it's just annoying.
This patch replaces usage of tempname with mkstemp, avoiding the
deprecation warning. It also removes the temporary files when the test
is complete, which apparently we weren't doing before (oops).
Review: https://reviewboard.asterisk.org/r/2957/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was
replaced with one of my own.
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909)
clicompat-r2.patch uploaded by coreyfarrell (license 5909)
codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909)
main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 401704 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 401705 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
slightly different than my proposal. Instead of putting an 'extends'
field on the subtype, the base type has a 'subTypes' field, which is a
list of the subTypes. Given that its a messaging model and not an
object model, kinda makes sense.
This patch changes the events.json api-doc, and the python translators
to take the new format into account.
Other changes that are in Swagger 1.2 were not adopted, since the spec
is still in flux, and could change before it's finalized.
A summary of changes to the Swagger-1.2 spec can be found at
https://github.com/wordnik/swagger-core/wiki/1.2-transition.
(closes issue ASTERISK-22440)
Review: https://reviewboard.asterisk.org/r/2909/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses
was removed by mistake. This patch restores this functionality and adds notes
about IPv6 addresses in the sample config.
(closes issue ASTERISK-22741)
Reported by: Joshua Colp
Tested by: Michael L. Young
Patches:
asterisk-22741-fix-binding-multiple-addr.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2945/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r400089, a patch was put in to correct erroneous RTCP statistic resets.
Unfortunately, ast_rtp_read can be called on an RTP instance that does not
have RTCP information. This patch prevents that crash by only resetting
the statistics if we do actually have an RTCP instance.
(issue AST-1174)
(closes issue ASTERISK-22667)
Reported by: John Bigelow
........
Merged revisions 401445 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 401446 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The queue_log entry resulting from CLI "queue remove member" when
log_membername_as_agent is enabled is wrong. It always uses the interface
name instead of the member name in the queue_log entry.
* Get the queue member before removing it from the queue so the member
name is available for the queue_log entry.
(closes issue ASTERISK-21826)
Reported by: Oscar Esteve
Patches:
fix_membername.diff (license #6505) patch uploaded by Oscar Esteve
(modified to fix potential ref leak)
........
Merged revisions 401433 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original issue noted that the bridge is orphaned when res_parking.so
is not loaded and a call uses the dial kK flags.
A similar issue happens when only one of the park flags is used. In this
case you have the bridge with one or the other channel left in it. The
channel and bridge will stay around until the channel hangs up.
* Fixed the initial bridge channel push failure to act as if the channel
were kicked out of the bridge. The bridge then decides if it needs to be
dissolved.
(closes issue ASTERISK-22629)
Reported by: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/2928/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Parking timeouts did not set any DTMF features for the channel calling the
parker back.
* Added code to set the parkedcalltransfers, parkedcallreparking,
parkedcallhangup, and parkedcallrecording options appropriately for the
channels when a parking timeout occurs. The recall channel DTMF options
are set using the BRIDGE_FEATURES channel variable to allow the other
timeout options to have the DTMF features available.
(closes issue ASTERISK-22630)
Reported by: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/2942/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Updated the XML documentation to indicate that the parkedcalltransfers,
parkedcallreparking, parkedcallhangup, and parkedcallrecording
configuration options also apply to parking timeouts.
(issue ASTERISK-22630)
Reported by: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/2942/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
chan_dahdi: Fix unable to get index warning when transferring an analog call.
Transferring an analog call using flashhooks generated an unable to get
index WARNING message when the transfer is completed.
* Removed unnecessary analog subchannel shell games when transferring a
call using flashhooks.
Thanks to Tzafrir Cohen for mentioning this in a comment on issue
ASTERISK-22720.
........
Merged revisions 401378 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This particular debug message, during a stress test, was logged so
often that it appeared that there may be a memory leak in the logger
code. In actuality, there was no memory leak, but the logger thread
was having a hard time keeping up with the demands of the rest of the
system.
Since this debug message has no value at all, the best way to fix the
problem was to just remove the message.
(closes issue AST-1225)
reported by John Bigelow
Patches:
spammy_log.diff uploaded by Mark Michelson (License #5049)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes address lookup for incoming calls without a peer definition.
The address family was unset instead of being set to AST_AF_UNSPEC
which was causing lookup failures on "127.0.0.1". This is one of the
causes of the current failure of the app_page integration test.
Review: https://reviewboard.asterisk.org/r/2933/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change allows a user of ARI to know what channel it has originated and also follow any
progress. If a Stasis application is provided it will be automatically subscribed to the
originated channel immediately.
(closes issue ASTERISK-22485)
Reported by: David Lee
Review: https://reviewboard.asterisk.org/r/2910/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A common idiom in Asterisk is to due something like:
for (ao2_obj = list_beginning; ao2_obj = next_item; ao2_ref(ao2_obj, -1)) {
...do stuff...
}
This is nice because it automatically takes care of the object references
for you. However, there is a pitfall here. If a break statement is in the
for loop, then the current reference is not cleaned up. In some cases, this
is on purpose, but in others there is a leak. This commit fixes the leak
cases.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most callers of ast_channel_make_compatible() happen before the channels
enter a two party bridge. With the new bridging framework, two party
bridging technologies may also call ast_channel_make_compatible() when
there is more than one thread involved with the two channels.
* Added channel lock protection in set_format() and
ast_channel_make_compatible_helper() when dealing with the channel's
native formats while setting up a translation path.
* Fixed best_src_fmt and best_dst_fmt usage consistency in
ast_channel_make_compatible_helper(). The call to
ast_translator_best_choice() got them backwards.
* Updated some callers of ast_channel_make_compatible() and the function
documentation. There is actually a difference between the two channels
passed in.
* Fixed the deadlock potential in res_fax.c dealing with
ast_channel_make_compatible(). The deadlock potential was already there
anyway because res_fax called ast_channel_make_compatible() with chan
locked.
(closes issue ASTERISK-22542)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2915/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When trying to determine if a peer is behind NAT, we should not be using the
ports when comparing addresses.
This patch removes the port from being checked and just useds the addresses
now.
(closes issue ASTERISK-22729)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-remove-using-port-for-nat-check.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2927/
........
Merged revisions 401182 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
