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Currently, when the first marked user enters the conference that
contains waitmarked users, a prompt is played indicating that the user
is being placed into the conference. Unfortunately, this prompt is
played to the marked user and not the waitmarked users which is not
very helpful.
This patch changes that behavior to play a prompt stating
"The conference will now begin" to the entire conference after adding
and unmuting the waitmarked users since the design of confbridge is not
conducive to playing a prompt to a subset of users in a conference in
an asynchronous manner.
(closes issue PQ-1396)
Review: https://reviewboard.asterisk.org/r/3155/
Reported by: Steve Pitts
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When formatting an optional IE, the value is, of course, optional. As such, it
is entirely appropriate for ast_json_object_get to return NULL. If that occurs,
we now simply skip the IE that was requested, as it was not provided by the
entity that raised the event.
Thanks to George Joseph (gtjoseph) for catching this and reporting it in
#asterisk-dev
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This change allows timing implementation data to be stored directly
on the timer itself thus removing the requirement for many
implementations to do a container lookup for the same information.
This means that API calls into timing implementations can directly
access the information they need instead of having to find it.
Review: https://reviewboard.asterisk.org/r/3175/
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When extracting timestamps that are parsed, time stamp values that are not set
(time values of 0.000000) should not actually result in a parsed string. The
value should be skipped, and the result of the CDR function should be an
empty string.
Prior to this patch, the result was fed to the time formatting, which would
result in an output of a date/time in 1969.
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Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect. The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.
For example:
1) v1.4 calls v1.8 (or later) using IAX2
2) v1.8 answers and sends a connected line update control frame. (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)
3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)
4) v1.4 disconnects the call once the receive queue becomes empty.
Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:
* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.
* Made block sending and receiving control frames that have no reason to
go over the wire.
* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.
* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.
(closes issue AST-1302)
Review: https://reviewboard.asterisk.org/r/3174/
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The appdocsxml.dtd specifies that a "required" attribute in a parameter may
have a value of yes, no, true, or false. On some systems, specifying "False"
instead of "false" would cause a validation error. This patch fixes the casing
to explicitly match the DTD.
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* If the "stutter" (voicemail indication) tone is indeed a stutter tone,
and it ends with a constant tone, make sure that it is the dial tone.
This was done for India (in), Mexico (mx) and the Philippines (ph).
* If no "stutter" tone exists for a country, provide one. This was done for
Spain (es), Malaysia (my) and Venezuela (ve).
Review: https://reviewboard.asterisk.org/r/3158/
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This patch adds documentation for the Security Events that are emited over
AMI. It also notes these events in the UPGRADE/CHANGES file.
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needs a little clarification
There is a bit of nuance to how you name things in pjsip.conf. This is a documentation patch to at least clear it up a little for users.
Review: https://reviewboard.asterisk.org/r/3180/
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If an enum had been previously created the alembic script would attempt to
re-create it and an error would be generated while running migrations for a
postgresql server. The work around for this is to use the ENUM object type
for postgres as opposed to the generic enum type used by sqlalchemy. Using
this type in the script seems to work properly for both postgres and mysql.
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* Adds identify, transport, and registration support to the PJSIP CLI.
* Creates three additional callbacks, one for an iterator, one for a
comparator, and one for a container. This eliminates the link dependency
from higher level modules to lower level ones.
* Eliminates duplicate sorting in PJSIP CLI commands.
* Cleans up PJSIP CLI output formatting.
* Pushes CLI command registration down to the implementing source file.
* Adds several ast_sip_destroy_sorcery functions to complement existing
ast_sip_sorcery_initialize functions. The destroy functions unregister
PJSIP CLI commands and PJSIP CLI formatters.
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3104/
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what is supported
Modifying the log message to be more specific as to what is supported. Specifically it seems format_wav supports only PCM encoded versions with a lower-case '.wav' extension.
(closes issues ASTERISK-22310)
Reported by: Jim Credland
Review: https://reviewboard.asterisk.org/r/3188/
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The changes log was written with language that was a little too internal
Asterisk specific, so it's been changed to be more in the frame of reference
of an ARI user. Also, previously the AMI event changes were omitted from the
change log as well as the ability to include a bridge name in the ARI post
bridges command.
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This fixes path handling for log files so that an extra / is not
appended to the file path when the path is absolute (begins with /).
This would previously result in different but functionally equivalent
paths in the output of 'logger show channels'.
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If the global section was not specified in pjsip.conf then the configuration
object does not exist in sorcery so when retrieving "debug" option it would
return NULL. Then the NULL result was passed to ast_false utils function
which would return false because it wasn't set to some representation of
false, thus enabling sip debug logging. Made it so if the global config object
does not exist then it will return a default of "no" for sip debugging.
(issue ASTERISK-23038)
Reported by: Rusty Newton
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Adds note of additional 0 for operator option on app_record
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Due to backwards compatible changes made to AMI/ARI, the version needs to
be bumped to 1.1.0/2.1.0, respectively.
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consistent.
Nothing actually cares about the value anyway.
(closes issue ASTERISK-23178)
Reported by: Jonathan Rose
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(closes issue ASTERISK-23168)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3143/
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Thanks to Guillaume Martres for doing the necessary research to validate
the change.
(closes issue ASTERISK-17727)
Reported by: LN
Patches:
use_certificate_chain.patch (license #5864) patch uploaded by st
documente_certificate_chain.patch (license #6576) patch uploaded by Guillaume Martres
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Thanks to snuffy for pointing this issue out and fixing it.
(closes issue ASTERISK-23250)
Reported by: snuffy
patches:
func_cdr-fix.diff uploaded by snuffy (License 5024)
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use.
The code assumed that unregistering the alias would always succeed while in
practice this is not actually true. A common case is the "reload" command itself.
If the cli_aliases.conf configuration file was changed and reload executed the
command would fail to unregister and ultimately point to freed memory.
The reload process now checks whether unregistering succeeded or not and if not
the old CLI alias is retained.
(closes issue ASTERISK-19773)
Reported by: Joel Vandal
(closes issue ASTERISK-22757)
Reported by: Gareth Blades
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Locking issues in skinny when picking up a call that doesn't exist. Cleaned
up sub locking by fully removing and using the chan lock instead. Also
changed ast_call_pickup to check whether chan was masq'd.
(closes issue ASTERISK-23249)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-locking01.diff uploaded by wedhorn (license 5019)
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This patch brings CDR processing further in line with r407085. During some dial
operations, the application would not be locked to the Dial application and
would instead continue to show the previously known application. In particular,
this would occur when a Parked call would time out. This was due to a previous
snapshot already locking the application to Park - processing this in a Dial
Begin allows the Dial application to reassert its rightful place.
(CDRs. Ugh.)
But hooray for the Parked Call tests for catching this in the Asterisk Test
Suite.
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This change enables transfers within ARI created bridges and adds events
for when they occur. Unlike other events these will be received if *any*
subscribed object is involved in the transfer.
(closes issue ASTERISK-22984)
Reported by: David M. Lee
Review: https://reviewboard.asterisk.org/r/3120/
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STACK_PEEK requires 2 parameters and LOCAL_PEEK requires 1 parameter. This
protects against situations where those parameters are blank or missing by
logging an error and returning.
(closes issue ASTERISK-23220)
Reported by: James Sharp
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This patch fixes a number of small-ish problems that were noticed when
witnessing the records that the FreePBX dialplan produces:
(1) Mid-call events (as well as privacy options) have the ability to change the
overall state of the Dial operation after the called party answers. This
means that publishing the DialEnd event when the called party is premature;
we have to wait for the execution of these subroutines to complete before
we can signal the overall status of the DialEnd. This patch moves that
publication and adds handlers for the mid-call events.
(2) The AST_FLAG_OUTGOING channel flag is cleared if an after bridge goto
datastore is detected. This flag was preventing CDRs from being recorded
for all outbound channels that had a 'continue' option enabled on them by
the Dial application.
(3) The CDR engine now locks the 'Dial' application as being the CDR
application if it detects that the current CDR has entered that app. This
is similar to the logic that is done for Parking. In general, if we entered
into Dial, then we want that CDR to record the application as such - this
prevents pre-dial handlers, mid-call handlers, and other shenaniganry
from changing the application value.
(4) The CDR engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more places
to determine if the channel is in hangup logic or dead. In either case, we
don't want to record changes in the channel.
(5) The default option for "endbeforehexten" has been changed to "yes". In
general, you don't want to see CDRs in the 'h' exten or in hangup logic.
Since the semantics of that option changed in 12, it made sense to update
the default value as well.
(6) Finally, because we now have the ability to synchronize on the messages
published to the CDR topic, on shutdown the CDR engine will now synchronize
to the messages currently in flight. This helps to ensure that all
in-flight CDRs are written before shutting down.
(closes issue ASTERISK-23164)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3154
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The parsing for the destination of the macro/gosub uses the '^' character to
separate out context, extension, and priority. However, the logic for the
macro/gosub execution was written such that it would only do the actual
macro/gosub jump if a '^' character existed. This doesn't apply when the
macro/gosub jump occurs in a priority/priority label. This patch changes
the logic so that the parsing still occurs, but the jump will occur even
for priorities/priority labels.
(issue ASTERISK-23164)
Review: https://reviewboard.asterisk.org/r/3154
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Added a "debug" configuration option for res_pjsip that when set to "yes"
enables SIP messages to be logged. It is specified under the "system" type.
Also added an alembic script to add the option to realtime.
(closes issue ASTERISK-23038)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3148/
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Removed the exportation of global symbols from the module as it is no longer
needed and it could potentially cause load problems as on some systems it
would try to load before res_pjsip_pubsub
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* Made ChanSpy accept a channel uniqueid or a fully specified channel name
as the chanprefix parameter if the 'u' option is specified.
(closes issue AFS-42)
Review: https://reviewboard.asterisk.org/r/3160/
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When the PJSIP pubsub framework was created, subscription handlers were required
to state what event they handled along with what body types they knew how to
generate. While this serves well when implementing a base RFC, it has problems
when trying to extend the body to support non-standard or proprietary body
elements. The code also was NOTIFY-specific, meaning that when the time comes
that we start writing code to send out PUBLISH requests with MWI or presence
bodies, we would likely find ourselves duplicating code that had previously been
written.
This changeset introduces the concept of body generators and body supplements. A
body generator is responsible for allocating a native structure for a given body
type, providing the primary body content, converting the native structure to a
string, and deallocating resources. A body supplement takes the primary body
content (the native structure, not a string) generated by the body generator and
adds nonstandard elements to the body. With these elements living in their own
module, it becomes easy to extend our support for body types and to re-use
resources when sending a PUBLISH request.
Body generators and body supplements register themselves with the pubsub core,
similar to how subscription and publish handlers had done. Now, subscription
handlers do not need to know what type of body content they generate, but they
still need to inform the pubsub core about what the default body type for a
given event package is. The pubsub core keeps track of what body generators and
body supplements have been registered. When a SUBSCRIBE arrives, the pubsub core
will check that there is a subscription handler for the event in the SUBSCRIBE,
then it will check that there is a body generator that can provide the content
specified in the Accept header(s).
Because of the nature of body generators and supplements, it means
res_pjsip_exten_state and res_pjsip_mwi have been completely gutted. They no
longer worry about body types, instead calling
ast_sip_pubsub_generate_body_content() when they need to generate a NOTIFY body.
Review: https://reviewboard.asterisk.org/r/3150
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A couple of the scripts had errors that would not allow a full migration to
take place. The extensions table needed to make its 'id' column a primary
key in order to work with mysql. The other script ...add_endpoints... was
missing tables that it was trying to add columns to.
Added the primary key on id for extensions and added the tables in for the
missing pjsip configuration options. While it is not ideal to modify already
released scripts this was a case where it had to be done due to errors in
the script and lacking a better alternative.
Review: https://reviewboard.asterisk.org/r/3167/
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When subscribing to MWI (res_pjsip_mwi) and the sip uri did not contain a name
(ex: sip:<ip address>) then the subscription would fail since it would be unable
to locate an associated aor. This patch makes it so that when a subscribe comes
with no aor name then it will subscribe to all aors on the located endpoint.
(closes issue ASTERISK-23072)
Reported by: Bob M
Review: https://reviewboard.asterisk.org/r/3164/
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In NAT scenarios where a call is placed to a Grandstream phone,
res_pjsip will sometimes send the ACK to a 200 OK to the private
address of the device behind the NAT instead of the address of the NAT
device. This corrects that behavior by rewriting the address in the
Contact header in the incoming 200 OK and the dialog's target address
if necessary (since it has already been rewritten to the incorrect
private address).
(closes issue ASTERISK-23106)
Review: https://reviewboard.asterisk.org/r/3168/
Reported by: Matt Jordan
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Return before chan is possibly unlocked a second time when hanging up
a channel in SUBSTATE_OFFHOOK.
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ast_bind to a port reserved for another program by SELinux causes
errno == EACCES. This caused random failures when binding rtp or
udptl sockets. Treat EACCES as a non-fatal error, try next port.
(closes issue ASTERISK-23134)
Reported by: Corey Farrell
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Closes issue ASTERISK-22662
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What seems to be happening is if a subscription has been terminated and the
subscription timeout/expires is less than the time it takes for all pending
transactions (currently on the subscription) to end then the subscription
timer will not have been canceled yet and sub will be null. Since the
subscription has already been canceled nothing needs to be done so a null
check in the asterisk code is sufficient in working around this problem.
(closes issue ASTERISK-23129)
Reported by: Dan Jenkins
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Asterisk's RADIUS module currently build against libradiusclient-ng, but this
project has been superseeded by libfreeradius-client. The API is 99% compatible
except that the header name has changed, the library name has changed, and
the configuration file location has changed.
(closes issue ASTERISK-22980)
Reported by: Jeremy Lainé
Patches:
freeradius-client.patch uploaded by sharky (license 6561)
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On some systems the values for INFINITY and NAN are not defined thus causing
a build error on those systems. Added definitions for those if they had
not previously been defined.
(closes issue ASTERISK-23056)
Reported by: capouch
Patches:
inf-nan-patch.txt uploaded by capouch (license 6564)
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Currently, attempting to subscribe an application to a device state
that it has already subscribed to will generate a 500 error response.
This will now be treated as a subscription refresh even though ARI
subscriptions don't currently support lifetimes and will respond with
the normal response for a successful subscription (200 OK).
(closes issue ASTERISK-23143)
Reported by: Matt Jordan
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In ast_rtp_instance_make_compatible(), after a failure of
channel tech call get_rtp_info() to return peer_instance,
the null pointer would be passed to ao2_ref, producing an
error that looked like a refernce counting problem but is
not. This patch corrects that and adds helpful LOG_ERROR
messages to indicate which failure path occurred.
(issue AST-1276)
Review: https://reviewboard.asterisk.org/r/3156/
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* Fixed the test_cel_attended_transfer_bridges_link unit test to also
account for the local channel link being destroyed now that the bridges
are actually destroyed.
* Made CDR unit test use its own version of do_sleep() from the CEL unit
tests.
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Added a note in the changes file about the new 'StatusText' field that was
added to the 'ExtensionStatus' event.
(issue ASTERISK-23154)
Reported by: Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When an 'ExtensionStatus' event was raised it included the status as a
numerical value, but did not include a text description of the status.
Added a 'StatusText' field to the event which is a string representation
of the extension status. Also added this to the 'Extension State' command
response.
(closes issue ASTERISK-23154)
Reported by: Jonathan Rose
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