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using a regex.
Review: https://reviewboard.asterisk.org/r/2332/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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media can and will come from different places.
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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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This was causing build failures against external libraries that happened to use
them, unless silly hacks were added to the modules that used those headers.
Review: https://reviewboard.asterisk.org/r/2359/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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is forked.
(closes issue ASTERISK-20638)
Reported by: eelcob
Patches:
pedantic-call-pickup-from-tag.patch uploaded by eelcob (license 6442)
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* The powerpcspe Linux port uses linux-gnuspe as the OS string.
* Our build system shouldn't really care for that, so just call it linux-gnu.
* Original report: Roland Stigge , http://bugs.debian.org/701505
Review: https://reviewboard.asterisk.org/r/2357/
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Parsing Remote-Party-ID will now succeed if display-name is of the
*(token LWS) kind and not just the quoted-string kind.
Review: https://reviewboard.asterisk.org/r/2341/
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As of r380522 the configure scripts converts the value of linux-gnueabi*
of OSARCH to "linux-gnu". So no point in testing for those values.
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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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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
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Adds both fixed and variable prinotify messages and clearprinotify messages to skinny.
Also adds cli function for pushing messages to devices. i
Initial code by snuffy, expanded by myself to include fixed messages.
(closes issue ASTERISK-21091)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-prinotify02.diff uploaded by wedhorn (license 5019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Somehow, chan_jingle has managed to operate for years without setting the
sin_family on its bindaddr socket. This patch properly sets the field during
initial module load to AF_INET.
Note that the patch on the issue was modified slightly to change the
initialization of the socket from allocation of a chan_jingle private to the
module initialization, as the bindaddr object (which is static) only needs to
have the address set once.
(closes issue ASTERISK-19341)
Reported by: andre valentin
patches:
0105-chan_jingle.patch uploaded by avalentin (License 6064)
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When converting AMI class authorizations to a string representation, the
method always appends the ALL class authorization. This is especially
important for events, as they should always communicate that class
authorization - even if the event itself does not specify ALL as a class
authorization for itself. (Events have always assumed that the ALL class
authorization is implied when they are raised)
Unfortunately, this did mean that specifying a user with restricted class
authorizations would show up in the 'manager show user' CLI command as
having the ALL class authorization.
Rather then modifying the existing string manipulation function, this patch
adds a function that will only return a string if the field being compared
explicitly matches class authorization field it is being compared against.
This prevents ALL from being returned unless it is actually specified for
the user.
(closes issue ASTERISK-20397)
Reported by: Johan Wilfer
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The ParkAndAnnounce application documentation for the optional return_context
parameter states the following:
return_context
The goto-style label to jump the call back into after timeout. Default
'priority+1'.
Unfortunately, the application was sending the channel back into the dialplan
at 'priority', which is the ParkAndAnnounce application call. This causes an
infinite loop of the channel constantly being parked, announced, timed out,
parked, announced, timed out... while fun, especially for those callers you
wish to drive to the end of madness, this was not the intent of the
application.
(closes issue ASTERISK-20113)
Reported by: serginuez
patches:
app_parkandannounce.diff uploaded by serginuez (License 6405)
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When IPv6 support was added to FastAGI, the intent was to have the ability to
check all addresses resolved for a host since we might receive an IPv4 address
and an IPv6 address. The problem with the current code, is that, since we are
doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring
this instead of handling it. We break out of the loop and continue on. When we
later call ast_poll(), it succeeds but we never check if we have a connection or
not on the socket level. We then attempt to send data to the host address that
we think is setup and it fails. We then check the errno and see that we have
"connection refused" and then return with agi failed.
This patch does the following:
* Handles EINPROGRESS by creating the function handle_connection()
- ast_poll() was moved into this function
- This function checks the results of the connection on the socket level after
calling ast_poll()
* Continues to the next address if the above fails to create a connection
* Once all addresses resolved are tried and we still are unable to establish a
connection, then we return that the FastAGI call failed
(closes issue ASTERISK-21065)
Reported by: Jeremy Kister
Tested by: Jeremy Kister, Michael L. Young
Patches:
asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2330/
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Apparently this feature became broken in 11, probably as a result
of the Hangup Cause project.
(closes issue ASTERISK-21113)
Reprted by: Heiko Wundram
Patches:
app_dial.patch uploaded by Heiko Wundram (license 5822)
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Asterisk was a little too pro-active in claiming that it found launchd. On
systems without launchd - such as FreeBSD - this resulted in certain items
in Asterisk that conflict with launchd to not be selectable, such as
res_timing_kqueue.
(closes issue ASTERISK-20749)
Reported by: Oleg Baranov
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The incorrect callid was being written to the "data1" field in queue_log table
for transfer events. The callid of the queue was being written instead of the
transfer target's callid. This now gets the correct "transfer to" number and
places that in the "data1" field of the queue_log table when a transfer event
is triggered.
(closes issue ASTERISK-19960)
Reported by: vladimir shmagin
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When a module's configuration is not loadable, we still load the module but it
is not in a running state. When trying to troubleshoot, let's say, why
chan_motif is ignoring inbound XMPP traffic, there is no way to indicate that a
loaded module is not currently running.
(closes issue ASTERISK-21108)
Reported by: Rusty Newton
Tested by: Michael L. Young
Patches:
asterisk-21108_add_status-v2.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2331/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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If you started/stopped recording of a conference multiple times channels
would remain active even when all participants left the conference. This
was due to the fact that a reference to the confbridge was being added
every time a start record command was issued, but when the recording was
stopped there was no matching de-reference thus keeping the conference alive.
Made sure only a single reference is added for the record thread no matter how
many times recording is started/stopped. A de-reference is issued upon thread
ending.
Note, this issue is being fixed under AST-1088 since it relates to it and
should have been corrected along with those modifications.
(issue AST-1088)
Reported by: John Bigelow
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Currently, if one starts, stops, and then starts a recording again for a
conference the recorded data is appended to the file originally created
on the first record start. An option record_file_append has been added
that defaults to "yes", but when set to "no" will force creation of a new
file between every record start/stop.
(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/
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Patch adds all the packet and structure stuff to skinny to enable setting
service URLs in skinny, such as corporate directories.
This stuff is only relevant during load/unload as when activated. Also
some minor changes removing duplicated counting of addons and speedials in
handle_skinny_show_devices.
Review: https://reviewboard.asterisk.org/r/2321/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Auto complete for skinny debug allows multiple options and negation, also add
debug all option. Usage example: 'skinny debug all -packets' (each can be
autocompleted including -packet).
Change show device to use device name. Remove the duplicate ast_strdup's from
place calling device complete return immediately from complete devicename and
complete linename so that multiple options are displayed on the CLI if more
than one option available.
Review: https://reviewboard.asterisk.org/r/2333/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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A deadlock occurred after starting/stopping and then restarting a confbridge
recording. Upon starting a recording a record thread is created that holds a
lock until just before exiting. Stopping the recording does not stop/exit the
thread or release the lock. The thread waits until recording begins again.
Starting a stopped recording signals the thread to continue and start recording
again. However restarting the recording also created another record thread
resulting in a deadlock. The fix was to make sure the record thread was only
created once.
Also it was noted that filenames for the recordings were being concatenated for
each start/stop. This was fixed by creating a new file for each conference
session and appending the actual recorded data within the file (e.g. passing
the 'a' option to MixMonitor).
(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/
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The "registertrying" option was removed in r343220. The "rtp_engine"
option was added in r186078 but erroneously named "engine" in the sample.
Note that there is no global sip setting for a different engine.
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Notes that the 'e' option actually decodes data when used as a write function
such as with the SET application while it encodes data when used to read.
Review: https://reviewboard.asterisk.org/r/2335/
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* Added the following flags to the CLI "confbridge list <conference>" output:
A - The user is an admin
M - The user is a marked user
W - The user must wait for a marked user to join
E - The user will be kicked after the last marked user leaves the conference
w - The user is waiting for a marked user to join
* Added the following header to the AMI ConfbridgeList events:
WaitMarked, EndMarked, and Waiting.
(closes issue AST-1101)
Reported by: John Bigelow
Patches:
confbridge-show-admin3.txt (license #5091) patch uploaded by John Bigelow
Modified
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Previously, presencestate information was sent whenever the state was not
NOT_SET. When r381594 actually returned INVALID presence state in all the
places it was supposed to, it caused chan_sip to start adding presence
state information to NOTIFY requests that it previously would not have
added. chan_sip shouldn't be adding presence state information when the
provider is in an invalid state; users can't set the state to invalid and
an invalid state always implies that the provider is in an error condition.
(issue AST-1084)
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their id.
Review: https://reviewboard.asterisk.org/r/2329/
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This patch fixes a crash in Asterisk that could be caused by using the
PresenceState AMI action while providing an invalid provider. This patch
also adds some additional warnings when a user attempts to provide the
PresenceState action with invalid data, and removes some NOTICE statements
that were still lurking in the code from testing.
(closes issue AST-1084)
Reported by: John Bigelow
Tested by: John Bigelow
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Reference counting for the channel and its tech_pvt got messed up at
some point between 1.8 and 11. The result was that if a BYE for a dialog
that had been replaced (via an INVITE with Replaces) was received, Asterisk
would crash due to trying to access data on a channel that was no longer there.
The fix I introduced is to remove code that both unrefs the sip_pvt and sets
the channel's tech_pvt to NULL when an INVITE with Replaces is handled. This
way when a BYE is received, the tech_pvt will be non-NULL and so the BYE can
be processed and not cause a crash.
(closes issue ASTERISK-20929)
reported by Kristopher Lalletti
patches:
ASTERISK-20929.patch uploaded by Mark Michelson (License #5049)
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This patch does two things:
1. It disables (temporarily) strict XML documentation checking for module
configurations. We should re-enable it before making any release from
trunk.
2. Pass the module flag AST_MODULE through sorcery. This means several of the
API calls are now macros and will do this automatically for you. The config
framework needs the module that objects are registering to so it can
properly construct the documentation. (This was already a required field,
but sorcery was getting by without it)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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While autoservice is running and servicing a channel the callid is being stored
and removed in the thread's local storage for each iteration of the thread loop.
If debug was set to a sufficient level the log file would be spammed with callid
thread local storage debug messages.
Added a new function that checks to see if the callid to be stored is different
than what is already contained (if anything). If it is different then
store/replace and log, otherwise just leave as is. Also made it so all logging
of debug messages pertaining to the callid thread storage outputs only when
TEST_FRAMEWORK is defined.
(issue ASTERISK-21014)
(closes issue ASTERISK-21014)
Report by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2324/
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Some bad copy/pasting resulted in using the audio crypto attribute for both
text and video RTP. Also the audio crypto isn't set until after these, so it
was really just bad all around.
(closes ASTERISK-20905)
Reported by: Kristopher Lalletti
patches:
rtp_crypto_video_text.diff uploaded by Jonathan Rose (license 6182)
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This patch allows a module to define its configuration in XML in source, such
that it can be parsed by the XML documentation engine. Documentation is
generated in a two-pass approach:
1. The documentation is first generated from the XML pulled from the source
2. The documentation is then enhanced by the registration of configuration
options that use the configuration framework
This patch include configuration documentation for the following modules:
* chan_motif
* res_xmpp
* app_confbridge
* app_skel
* udptl
Two new CLI commands have been added:
* config show help - show configuration help by module, category, and item
* xmldoc dump - dump the in-memory representation of the XML documentation to
a new XML file.
Review: https://reviewboard.asterisk.org/r/2278
Review: https://reviewboard.asterisk.org/r/2058
patches:
on review 2058 uploaded by twilson
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Don't know why it seemed to work during testing, but it really is needed
for protocol v17 (and probably above).
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It doesn't hurt to check AST_SOFTHANGUP_UNBRIDGE either, but it should not
be set outside of a bridge.
(issue ASTERISK-20492)
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Fix chan_skinny so that it respects callerID presentation of inbound calls to
device and a couple of other minor fixes: 145 packet (add OCTAL_FROM amd callerid),
and dont send dialednumber message if protocol >= 17.
(closes issue ASTERISK-21066)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-respect-clid-restrictions-v2.diff uploaded by snuffy (license 5024)
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