If a peer is used in a register line and TLS is defined as the transport, the
registration fails since the transport on the dialog is never set properly
resulting in UDP being used instead of TLS.
This patch sets the dialog's transport based on the transport that was defined
in the register line. If the register line does not specify a transport, the
parsing function for the register line always defaults back to UDP.
(closes issue ASTERISK-21964)
Reported by: Doug Bailey
Tested by: Doug Bailey
Patches:
asterisk-21964-set-reg-dialog-transport.diff
by Michael L. Young (license 5026)
........
Merged revisions 396240 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 396248 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dial and Queue would previously apply a new set of features whenever
bridging. These options would be based purely on the options supplied
to the dial/queue applications. This patch changes the function those
applications use to bridge calls so that the features will be added
to the set of existing features for each channel rather than having
them override the existing features.
(closes issue ASTERISK-22209)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2713/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Suspend and unsuspend callbacks are added to the holding bridge so
that entertainment can be disabled and re-enabled when operations
would suspend a channel on the bridge (such as playback operations).
This fixes entertainment so that when those operations end, the
entertainment can pick back up and it also serves as an optimization.
Also, this patch fixes a bug caused by triggering ringing frames
immediately instead of pushing them to the queue which created a race
condition where sometimes parking with ringing during attended
transfers would cause the ringing to be interrupted by an unhold
frame.
(closes issue ASTERISK-22006)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2711/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396189 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Roles are now cleared with each entry into a bridge with addChannel.
If the roles parameter is present, the role specified will be applied
to all channels being added with the addChannel command.
(closes issue ASTERISK-21973)
Reported by: Matt Jordan
https://reviewboard.asterisk.org/r/2691/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds the following unit tests:
* create_lot: tests adding and removal of a new parking lot (baseline)
* park_extensions: creates a parking lot that registers extensions and
then confirms that all of the expected extensions exist
* extensions_conflicts: creates numerous parking lots to test that
extension conflicts in parking lots result in parking lot
creation failing
* dynamic_parking_variables: Tests that the creation of dynamic
parking lots respects the related channel variables set on the
channel that requests them.
* park_call: Tests adding a channel to a parking lot's holding bridge
by standard parking functions.
* retrieve_call: Tests pulling a channel out of a parking lot's
holding bridge via parked call retrieval functions.
(closes issue ASTERISK-22138)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2714/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the module
won't load on Linux.
This patch renames the variables, adding the ast_ prefix so they will
be exported.
Review: https://reviewboard.asterisk.org/r/2737
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AMI message router is owned wholly by manager.c. Previously, each of the
manager_{item} source files had their own message router and they unsubscribed
from each; once they moved over to using a single message router only a single
unsubscribe became necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds basic system information access to ARI.
The results are roughly what you get from 'core show settings', with a
few minor differences.
* Data is structured, with 'build', 'system', 'config' and 'status'
sub-objects.
* Each sub-object is selectable, using the ?only= parameter. A comma
separated list can be provided to select multiple sections.
* A few config options are numeric, for which 0 means 'unlimited'.
Instead of having a special interpretation of those fields, they
are simply omitted if they're 0.
* The information is limited to what might be useful to building
external applications.
(closes issue ASTERISK-21575)
Review: https://reviewboard.asterisk.org/r/2702/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Swagger allows parameters to be specified as 'allowMultiple', meaning
that the parameter may be specified as a comma separated list of
values.
I had written some of the API docs using that, but promptly forgot
about implementing it. This patch finally fills in that gap.
The codegen template was updated to represent 'allowMultiple' fields
as array/size fields in the _args structs. It also parses the comma
separated list using ast_app_separate_args(), so quoted strings in the
argument will be handled properly.
Review: https://reviewboard.asterisk.org/r/2698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* BridgeEnter now contains the unique ID of the channel that is to be swapped out, if applicable.
* There is a ParkedCallSwap event that is sent when a parked channel has a new channel take its place.
(closes issue ASTERISK-22193)
reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/2712
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This moves ast_str_container_alloc, ast_str_container_add,
ast_str_container_remove, and related private functions into
strings.c/h since they really don't belong in astobj2.c/h.
As a result of this move, utils also had to be updated.
Review: https://reviewboard.asterisk.org/r/2719/
(closes issue ASTERISK-22041)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is smaller than the initial review placed on review board. This is because
a change to allow for channel drivers to access parking functionality externally was
committed and invalidated quite a few of the changes initially made.
(closes issue ASTERISK-22039)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2717
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For chan_pjsip, this introduces CLI/AMI remote unregistration commands,
reworks CLI syntax for sending NOTIFYs, adds AMI qualification support,
and adds documentation for PJSIPNotify.
This also fixes two refcounting bugs in the outbound registration code.
Review: https://reviewboard.asterisk.org/r/2695/
(closes issue ASTERISK-21939)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It moves the pickup code out of features.c and into pickup.c
* It removes the vast majority of dead code out of features.c. In particular,
this includes the parking code.
(issue ASTERISK-22134)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some regex implementations won't compile an empty string. Assuming that
it's equivalent of a regex that will match anything, use ".?" instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds support for externally initiated parking requests. In particular,
chan_skinny has a protocol level message that initiates a call park.
This patch now supports that option, as well as the protocol specific
mechanisms in chan_dahdi/sig_analog and chan_mgcp.
* A parking bridge features virtual table has been added that provides
access to the parking functionality that the Bridging API needs. This
includes requests to park an entire 'call' (with little or no additional
information, thank you chan_skinny), perform a blind transfer to a parking
extension, determine if an extension is a parking extension, as well as the
actual "do the parking" request from the Bridging API.
* Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
functions
* The removal of some - but not all - dead parking code from features.c
This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)
Review: https://reviewboard.asterisk.org/r/2710
(closes issue ASTERISK-22134)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability in Queue to raise a hint when a member's paused
state changes. The hint uses the form 'Queue:{queue_name}_pause_{member_name}',
where {queue_name} and {member_name} are the name of the queue and the name
of the member to subscribe to, respectively.
For example: exten => 8501,hint,Queue:sales_pause_mark.
Members will show as In Use when paused.
Note that the format of the queue pause hint was changed slightly from what
is on the issue to accomodate suggestion on the code review.
Review: https://reviewboard.asterisk.org/r/2254
(closes issue ASTERISK-20842)
Reported by: Philippe Lindheimer
patches:
qpause-10-378206.diff uploaded by Philippe Lindheimer (license 5519)
qpause-11-378206.diff uploaded by Philippe Lindheimer (license 5519)
qpause-trunk-378206.diff uploaded by Philippe Lindheimer (license 5519)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.
To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.
In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:
single_topic ----------------> all_topic
^
|
single_topic_cached ----+----> all_topic_cached
|
+----> cache
This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.
Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.
(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch makes it so that all registration attempts that fail that
also permanently modify the registration state will raise an appropriate
AMI event.
Note that this patch was forward ported to trunk and the Stasis Core
message bus by mjordan.
(closes issue ASTERISK-21368)
Reported by: Dmitriy Serov
patches:
chan_sip.c.diff uploaded by Demon (license 6479)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows starting playback of audio through the CONTROL STREAM FILE
AGI command to start at a particular offset. It will also return the final
position of the file in the 'endpos' attribute.
(closes issue ASTERISK-17803)
Reported by: Murray Melvin
patches:
res_agi.c.r316293.diff uploaded by murraytm (license 6221)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This was created as a debugging tool before proper endpoint identifiers
were created. Using it now can actually lead to harmful results.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Michael L. Young
Jonathan Rose
Matthew Jordan
David M. Lee
Mark Michelson
Kinsey Moore
Joshua Colp