The sorcery astDB wizzard does not handle regex correctly if the pattern
begins with an anchor character.
This patch attempts to convert the anchored regex pattern to a prefix
pattern supported by astDB for performance reasons. If it is not able to
convert the pattern it falls back to getting all astDB members of the
family and doing a normal regex pattern matching on the retrieved records.
Review: https://reviewboard.asterisk.org/r/3161/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@408385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an endpoint sends a REGISTER request to Asterisk, we now will
associate the User-Agent header with all contacts that were bound in
that REGISTER request.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@408270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is highly unlikely, but - at least in Asterisk 12 - theoretically possible
to load Asterisk with no dialplan whatsoever. If that occurs, and some other
module (that is not a pbx module) attempts to merge its contexts into the
dialplan, the existing merge routine will crash. This is because it is not
insane, and rightly believes that you provided some sort of dialplan,
somewhere.
This patch will gracefully merge the contexts in such a case. Note that this
is highly unlikely to occur in 1.8/11, as features will most likely provide
some dialplan via parking. However, in Asterisk 12, parking is now provided
by res_parking, and hence may create its dialplan later.
(closes issue ASTERISK-23297)
Reported by: CJ Oster
Review: https://reviewboard.asterisk.org/r/3222
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@408220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
URI's are supposed to be case sensitive and all
lower case. In practice some portions of URI's
in ARI are case insensitive and others are not,
such as TECH, which in one instance would match
a lower case name and in another would not. In
this patch, the ast_endpoint_lastest_snapshot()
function is modified to change the TECH portion
to full upper case before lookup. This resolves
the discrepancy noted by the reporter. However
I chose to avoid forcing the /ari prefix of the
URI's to be lower case for now. Except for the
two cases here, all URI's should be lower case,
unless they are part of a resource name or id.
Review: https://reviewboard.asterisk.org/r/3211/
Reported by: Zane Conkle
(closes issue ASTERISK-23125)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@408140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ast_format_sdp_parse and ast_format_sdp_generate
the check checks for a valid interface and function
were potentially confusing, and hid an error in the
test of the presence of the function that is called
later. This patch clears up and corrects the test.
Review: https://reviewboard.asterisk.org/r/3208/
(closes issue ASTERISK-23098)
Reported by: marcelloceschia
Patches:
main_format.patch uploaded by marcelloceschia (license 6036)
ASTERISK-23098.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@408138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves setting SIP_DEFER_BY_ON_TRANSFER prior to calling
ast_bridge_transfer_blind. This prevents a BYE from being sent prior to the
NOTIFY request that informs the transferor if the transfer succeeded or failed.
This patch also clears said flag from the off nominal NOTIFY paths in the
local_attended_transfer code, as once we've sent the NOTIFY request it is safe
to send by the BYE request.
This was caught by the blind-transfer-accountcode test in the Asterisk Test
Suite.
(closes issue ASTERISK-23290)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3214/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@408069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
PJSIP has built-in MWI code that could be useful to some
degree, but our utilization of the API actually made our
code a bit more cluttered since we had to have special
cases peppered throughout.
With this change, we move to using the pjsip_evsub API
instead, which streamlines the code by removing special
cases.
Review: https://reviewboard.asterisk.org/r/3205
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@408005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an AOR has no permanent contacts, then the
permanent_contacts container is never allocated.
This makes the code safe in the face of NULLs.
I also changed the variable that counts contacts
from "num" to "total_contacts" since there are now
two variables that are indicate numbers of things.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The old code depended on undefined va_arg behaviour: calling a function
twice with the same va_list parameter and expecting it to continue where
it left off. The changed code behaves like the manpage says it should.
Also added a bunch of early returns to trap errors (e.g. OOM) instead of
crashing.
The problem was found by Julian Lyndon-Smith. The deviant behaviour on
the raspberry PI also uncovered another bug (fixed in r407875) in the
res_config_pgsql.so driver.
Reported by: jmls
Tested by: jmls
Review: https://reviewboard.asterisk.org/r/3201/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch tweaks the behaviour of POST /channels with channel variables such
that the variables are passed into the pbx.c routines that perform the
origination. This allows the variables to be assigned to the newly created
channels immediately upon their construction, as opposed to be assigned after
the originate has completed.
The upshot of this is that the variables are available on the channels if
they execute in the dialplan, as opposed to only being available once the
channels are answered.
Review: https://reviewboard.asterisk.org/r/3183/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds documentation for the Security Events that are emited over
AMI. It also notes these events in the UPGRADE/CHANGES file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When run in offline mode, this would attempt to check the database for
the presence of a type it was going to try to create. I now check the
context to see if we're running in offline mode and change a parameter
accordingly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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chan_local: Fix reversed LocalOptimization field in LocalBridge event
(closes issue ASTERISK-23232)
Reported by: Leon Roy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch updates the api.wiki.mustache template and the swagger_model python
script to understand if an operation has a body parameter. If an operation
does have a body parameter, it will now be displayed in the corresponding
wiki entry.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@407034 65c4cc65-6c06-0410-ace0-fbb531ad65f3