* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ast_filestream object gets tacked on to a channel via
chan->timingdata. It's a reference counted object, but the reference
count isn't used when putting it on a channel. It's theoretically
possible for another thread to interfere with the channel while it's
unlocked and cause the filestream to get destroyed.
Use the astobj2 reference count to make sure that as long as this code
path is holding on the ast_filestream and passing it into the file.c
playback code, that it knows it's valid.
Bug reported by Leif Madsen.
Review: https://reviewboard.asterisk.org/r/3135/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The core may (depending on circumstances) request a single codec on outgoing
calls. Many channel drivers ignore or treat this as a suggestion while still
including configured codecs. The res_pjsip_session logic treated this as
an explicit request, leaving out other configured codecs.
This change makes res_pjsip_session behave like other channel driver and simply
adds the requested codec to the list.
(closes issue ASTERISK-23082)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3140/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CEL data structures need to be protected during a configuration reload
and shutdown. Asterisk crashed during a shutdown because CEL events were
still in flight and the CEL data structures were already destroyed.
* Protected the cel_backends, cel_dialstatus_store, and cel_linkedids ao2
containers with a global ao2 object wrapper.
* Added NULL checks before use of the cel_backends, cel_dialstatus_store,
and cel_linkedids ao2 containers in case the CEL module is already
shutdown.
* Fixed overloading of the cel_linkedids held objects reference count.
During shutdown any held objects would be leaked.
* Fixed memory leak of cel_linkedids held objects if the LINKEDID_END is
not being tracked. The objects in the cel_linkedids container were not
removed if the LINKEDID_END event is not used.
* Added access protection to the cel_backends container during the CLI
"cel show status" command.
* Made cel_backends, cel_dialstatus_store, and cel_linkedids use the
standard ao2 callback templates for the hash and cmp functions.
* Eliminated unnecessary uses of RAII_VAR().
* Made ast_cel_engine_init() cleanup alocated resources on failure.
(closes issue AST-1253)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3128/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made register atexit shutdown routine only once in __init_manager().
* Fixed some initial load failure conditions in __init_manager().
* Made reset options to defaults on reload when the reload will actually
happen.
* Removed unnecessary container traversals of the white/black filters
during manager_free_user().
* ast_free() does not need a NULL check before calling.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic
number" error on a "core restart gracefully" command if an AMI connection
is established.
* Added ao2_global_obj protection to the sessions global container.
* Fixed the order of unreferencing a session object in session_destroy().
* Removed unnecessary container traversals of the white/black filters
during session_destructor().
(closes issue AST-1242)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3144/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When support for a realtime sorcery module was added in revision 386731, the
wrong property was accidentally used for setting the column name to be updated
in the database table. This patch fixes the typo.
(closes issue ASTERISK-23177)
Reported by: Denis
Tested by: Denis
Patches:
asterisk-23177-use-field-name.diff by Michael L. Young (license 5026)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* PIDF bodies were reporting an "open" state in many cases where
it should have been reporting "closed"
* XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
* SIP URIs in XPIDF bodies did not go through XML sanitization
* XML sanitization had some errors:
* Right angle bracket was being replaced with "&rt;" instead of ">"
* Double quote, apostrophe, and ampersand were not being escaped.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* PIDF bodies were reporting an "open" state in many cases where
it should have been reporting "closed"
* XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
* SIP URIs in XPIDF bodies did not go through XML sanitization
* XML sanitization had some errors:
* Right angle bracket was being replaced with "&rt;" instead of ">"
* Double quote, apostrophe, and ampersand were not being escaped.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Confbridge AMI and CLI commands for mute, unmute, and setting the
single video source can accept channel prefixes in lieu of a full
channel name, but documentation states only that it is required and is
a channel name. This corrects the documentation.
(closes issue PQ-1397)
Reported by: Steve Pitts
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Several of the playback error messages for invalid media input in
res_stasis_playback.c had the media name and channel name reversed.
They now correctly identify the channel name and media name.
Reported by: skrusty
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Making the help text for both more explicit regarding the format of mailbox identifiers. i.e. clarifying the format for app_voicemail mailboxes vs mailboxes from external MWI sources through modules such as res_external_mwi.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The PJSIP header parsing function (pjsip_parse_hdr) can generate more
than one header instance from a single header field. These header
instances exist as a list attached to the returned header and must be
handled appropriately when they are added to a message or else only the
first header instance will be used. This changes the linked list
functions used in outbound proxy code to merge the lists properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds back in support for specifying channel variables during an
originate without compromising the ability to specify query parameters
in the JSON body. This was accomplished by generating the body-parsing
code in a separate function instead of being integrated with the URI
query parameter parsing code such that it could be called by paths with
body parameters. This is transparent to the user of the API and
prevents manual duplication of code or data structures.
(closes issue ASTERISK-23051)
Review: https://reviewboard.asterisk.org/r/3122/
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@406003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Bad offset in reading second or more fragment of skinny packets. Fixed
to offset by char (single byte) rather than size of req.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Logging from the skinny session loop was providing some incorrect reasons
for exiting the loop. Cleaned up messages and handling so correct reason
displayed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Existing code would do a full device restart when "skinny reset device"
was entered at the CLI and do a reset when "skinny reset device restart"
entered.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds improvements to support for allow=all in
pjsip.conf so that it functions as intended. Previously,
the allow/disallow socery configuration would set & clear
codecs from the media.codecs and media.prefs list, but if
all was specified the prefs list was not updated. Then a
call would fail when create_outgoing_sdp_stream() created
an SDP with no audio codecs.
A new function ast_codec_pref_append_all() is provided to
add all codecs to the prefs list - only those not already
on the list. This enables the configuration to specify a
codec preference, but still add all codecs, and even then
remove some codecs, as shown in this example:
allow = ulaw, alaw, all, !g729, !g723
Also, the display order of allow in cli output is updated
to match the configuration by using prefs instead of caps
when generating a human readable string.
Finally, a change to create_outgoing_sdp_stream() skips a
codec when it does not have a payload code instead of the
call failing.
(closes issue ASTERISK-23018)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3131/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change implements support for HTTP Transfer-Encoding
chunked in both JSON and Form (post vars) body content. A
new function ast_http_get_contents() handles both regular
and chunked mode body, returning after the entire body is
received.
(closes issue ASTERISK-23068)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3125/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Made documentation more explicit as to the use of the both options.
(issue ASTERISK-23071)
(closes issue ASTERISK-23071)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an endpoint had previously dynamically registered a contact and the contact
information was successfully stored in astdb then upon restart the qualify
notifications would not be sent out if the qualify_frequency was set. This was
due to the fact that only permanent contacts were being checked and scheduled
for qualifies on startup. Modified the code to check and schedule all
registered contacts at startup.
(closes issue ASTERISK-23062)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3124/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
action_originate responds to the remote system with an error when cap==NULL,
but doesn't return (abort the originate). Patched to return.
(closes issue ASTERISK-23034)
Reported by: Corey Farrell
Patches:
ASTERISK-23034.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When path support was added and contacts were made available during
request creation and transmission, the code path used by outbound
qualify support was not modified correctly and was causing request
creation to fail. This ensures that outbound request creation with only
a contact and no dialog, endpoint, or uri can succeed which restores
qualify support.
Reported by: gtjoseph
Reported by: kharwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
According to the new standard for V.27 and V.32 they are able to transmit
at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be
updated to reflect this. Also, because of this change the default 'minrate'
value was updated to be 4800.
(closes issue ASTERISK-22790)
Reported by: Paolo Compagnini
Patches:
res_fax.txt uploaded by looserouting (license 6548)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When endpoints get loaded their device state gets set to 'INVALID' because the
channel driver has not been loaded yet. Fixed by updating the device state for
every endpoint upon load of the channel driver.
(closes issue ASTERISK-23065)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3123/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds Path support to chan_pjsip in res_pjsip_path.c with minimal
additions in res_pjsip_registrar.c to store the path and additions in
res_pjsip_outbound_registration.c to enable advertisement of path
support to registrars and intervening proxies.
Path information is stored on contacts and is enabled via Address of
Record (AoRs) and Registration configuration sections.
While adding path support, it became necessary to be able to add SIP
supplements that handled messages outside of sessions, so a framework
for handling these types of hooks was added in parallel to the
already-existing session supplements and several senders of
out-of-dialog requests were refactored as a result.
(closes issue ASTERISK-21084)
Review: https://reviewboard.asterisk.org/r/3050/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds the following AMI commands:
PUT mailboxes/mailboxName
modifies mailbox state and implicitly creates new mailboxes
GET mailboxes/mailboxName
retrieves a JSON representation of a single mailbox if it exists
GET mailboxes
retrieves a JSON array of all mailboxes
DELETE mailbox/mailboxName
deletes a mailbox
Note that res_mwi_external must be loaded for these functions to
actually do anything.
Review: https://reviewboard.asterisk.org/r/3117/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ASTERISK-12117, an improvement to insure consistant local from tags
on outbound registrations resulted in an undesirable behavior - caused
by leftover unexpired sip_pvt dialogs (with the previous cseq number),
resulting in many uncessary REGISTER requests. Instead of significant
rework of transmit_register(), this change deletes the dialogs after a
200 OK response indiciating a successful registration, keeping the old
dialogs from interfering with normal operation.
(closes issue ASTERISK-22946)
Reported by: Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/3109/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The per console verbose level feature as previously implemented caused a
large performance penalty. The fix required some minor incompatibilities
if the new rasterisk is used to connect to an earlier version. If the new
rasterisk connects to an older Asterisk version then the root console
verbose level is always affected by the "core set verbose" command of the
remote console even though it may appear to only affect the current
console. If an older version of rasterisk connects to the new version
then the "core set verbose" command will have no effect.
* Fixed the verbose performance by not generating a verbose message if
nothing is going to use it and then filtered any generated verbose
messages before actually sending them to the remote consoles.
* Split the "core set debug" and "core set verbose" CLI commands to remove
the per module verbose support that cannot work with the per console
verbose level.
* Added a silent option to the "core set verbose" command.
* Fixed "core set debug off" tab completion.
* Made "core show settings" list the current console verbosity in addition
to the root console verbosity.
* Changed the default verbose level of the 'verbose' setting in the
logger.conf [logfiles] section. The default is now to once again follow
the current root console level. As a result, using the AMI Command action
with "core set verbose" could again set the root console verbose level and
affect the verbose level logged.
(closes issue AST-1252)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3114/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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chan_sip: Hangup transferer/transferee when transfer to Parking fails
When performing a SIP transfer to a Park extension, if the Park fails, chan_sip
will currently not hang up either the transferer or the transfer target. This
results in the channels being orphaned with no thread to service frames,
resulting in stuck channels.
This patch immediately hangs up the two channels if a Park fails.
(closes issue ASTERISK-22834)
Reported by: rsw686
Tested by: rsw686
(closes issue ASTERISK-23047)
Reported by: Tommy Thompson
Tested by: Tommy Thomspon
Review: https://reviewboard.asterisk.org/r/3107
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Patch updates skinny devices with a SKINNY_CONNECTED callstate if an
inbound ringing or callwaiting call is answered elsewhere.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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res/Makefile: alias dist-clean to distclean
A 'make distclean' is equivalent to 'make dist-clean' in the top most Makefile.
This patch updates the res/Makefile to recognize both distclean and dist-clean.
Note that this is needed for removing build.mak, which can run into problems
if the source file of Asterisk or its path is changed after build.mak is
generated.
(issue ASTERISK-22480)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@405363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
