Age | Commit message (Collapse) | Author |
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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
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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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extconfig.conf was hard-coded to not allow nested includes for some reason.
The code has been this way since a patch was merged for ASTERISK-3333 (revision
4889), which was a significant update to this code ("Merge config updates").
I can't figure out any good reason why this should be limited. This patch just
removes the limit and uses the default nesting depth limit.
Closes issue ASTERISK-17837
Review: https://reviewboard.asterisk.org/r/3159/
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In pre-trunk versions (...12) it accepts a basic regex, which is
confusing because all other regexes in asterisk are of the
extended kind.
Review: https://reviewboard.asterisk.org/r/3147/
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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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live_ast can be used as a wrapper script to run asterisk, gdb or
valgrind. In those cases it runs them and returns the result. It is more
useful to use 'exec' to avoid having another odd process in the chain.
Review: https://reviewboard.asterisk.org/r/3110/
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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/
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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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This patch adds the LWP to core show locks output if it is available.
Review: https://reviewboard.asterisk.org/r/3142/
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* 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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Review: https://reviewboard.asterisk.org/r/3141/
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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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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)
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r406294 | mmichelson | 2014-01-23 15:00:24 -0600 (Thu, 23 Jan 2014) | 11 lines
Fix presence body errors found during testing:
* 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.
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r406295 | mmichelson | 2014-01-23 15:09:35 -0600 (Thu, 23 Jan 2014) | 11 lines
Fix presence body errors found during testing:
* 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.
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In ast_build_timing, initialize the timezone value to NULL
in order to avoid deferencing an uninitialized value later
when calling ast_destroy_timing. The timezone value could
be uninitialized if ast_build_timing were to fail due to a
zero length time string.
(closes issue ASTERISK-22861)
Reported by: Sebastian Murray-Roberts
Review: https://reviewboard.asterisk.org/r/3134/
Patches:
ast_build_timing-initialize-timezone.patch uploaded by coreyfarrell (license 5909)
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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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This change allows chan_sip to decline individual image streams over
unsupported transports in the SDP of the 200 response. Previously,
an image stream offer with RTP/AVP as the transport would cause
chan_sip to respond with a 488.
(closes issue ASTERISK-22988)
Reported by: adomjan
Original patch by: adomjan
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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
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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.
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Review: https://reviewboard.asterisk.org/r/3133/
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This restricts direct usage of global oseq so that all accesses are
locked and threads are not racing to get oseq values that they did not
claim.
This also fixes a build error in res_pktccops under dev mode.
(closes issue ASTERISK-23100)
Reported by: adomjan
Patch by: adomjan
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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.
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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
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Bad offset in reading second or more fragment of skinny packets. Fixed
to offset by char (single byte) rather than size of req.
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is valid.
* Also simplified some subddress handling code.
(closes issue ASTERISK-23008)
Reported by: Michael Cargile
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Logging from the skinny session loop was providing some incorrect reasons
for exiting the loop. Cleaned up messages and handling so correct reason
displayed.
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Previously PJSIP did not track hold/unhold and it would always simply be
'inuse'. This patch fixes that.
review: https://reviewboard.asterisk.org/r/3129/
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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.
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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/
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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/
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ASTERISK-23071
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issues 23061,23028,23046,23027
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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and aors
Made documentation more explicit as to the use of the both options.
(issue ASTERISK-23071)
(closes issue ASTERISK-23071)
Reported by: Matt Jordan
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Review: https://reviewboard.asterisk.org/r/3112/
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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/
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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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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
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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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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/
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Remove subversion conflict tag accidentally left in CHANGES
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The cel_manager module did not properly handle the case where the
configuration file was invalid. The module will now output a warning
message and disable itself if this occurs.
Reported by: Bryan Walters
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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/
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