The 'ari set debug' command has been enhanced to accept 'all' as an
application name. This allows dumping of all apps even if an app
hasn't registered yet. To accomplish this, a new global_debug global
variable was added to res/stasis/app.c and new APIs were added to
set and query the value.
'ari set debug' now displays requests and responses as well as events.
This required refactoring the existing debug code.
* The implementation for 'ari set debug' was moved from stasis/cli.{c,h}
to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted.
* In order to print the body of incoming requests even if a request
failed, the consumption of the body was moved from the ari stubs
to ast_ari_callback in res_ari.c and the moustache templates were
then regenerated. The body is now passed to ast_ari_invoke and then
on to the handlers. This results in code savings since that template
was inserted multiple times into all the stubs.
An additional change was made to the ao2_str_container implementation
to add partial key searching and a sort function. The existing cli
code assumed it was already there when it wasn't so the tab completion
was never working.
Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf
(cherry picked from commit 1d890874f3)
This change adds experimental support for providing RTCP
feedback information to codec modules so they can dynamically
change themselves based on conditions.
ASTERISK-26584
Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857
It was possible for a frame to be re-inserted into a jitter buffer after it
had been removed from it. A case when this happened was if a frame was read
out of the jitterbuffer, passed to the translation core, and then multiple
frames were returned from said translation core. Upon multiple frames being
returned the first is passed on, but sebsequently "chained" frames are put
back into the read queue. Thus it was possible for a frame to go back into
the jitter buffer where this would cause problems.
This patch adds a flag to frames that are inserted into the channel's read
queue after translation. The abstract jitter buffer code then checks for this
flag and ignores any frames marked as such.
Change-Id: I276c44edc9dcff61e606242f71274265c7779587
Function CHANNEL(rtcp,all_rtt) CHANNEL(rtcp,all_loss) CHANNEL(rtcp,all_jitter)
always return 0.0 due to wrong define of macro "AST_RTP_SATA_SET" and
"AST_RTP_STAT_STRCPY".
It should compare "combined" with "stat" not "current_stat".
ASTERISK-26710 #close
Reported-by: Aaron An
Tested-by: AaronAn
Change-Id: Id4140fafbf92e2db689dac5b17d9caa009028a15
Adds the ability for extensions to be registered to include filename and
line number so that dialplan show output can show the filename and line
number of a config file responsible for generating a given extension.
This only affects config modules that are written to use the new extension
registering functions. In this patch, that only includes pbx_config, so
extensions registered in extensions.conf and any included extension will
be shown in this manner. Extensions registered in this manner will show
the filename and line number *instead* of the registrar.
ASTERISK-26658 #close
Reported by: Jonathan R. Rose
Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30
The PJSIPShowRegistrationsInbound AMI command was just dumping out
all AORs which was pretty useless and resource heavy since it had
to get all endpoints, then all aors for each endpoint, then all
contacts for each aor.
PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail
events which meets the intended purpose of the other command and has
significantly less overhead. Also, some additional fields that were
added to Contact since the original creation of the ContactStatusDetail
event have been added to the end of the event.
For compatibility purposes, PJSIPShowRegistrationsInbound is left
intact.
ASTERISK-26644 #close
Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a
The latest Release candidate fails to create RTP streams when IPv6
is not available. Due to the changes made in September the ast_sockaddr
structure passed around to create these streams is always of AF_INET6
type, causing failure when used for IPv4. This patch adds a utility
function to check for availability of IPv6 and applies such check
at startup to determine how to create the ast_sockaddr structures.
ASTERISK-26617 #close
Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e
Use of the new logging is as simple as issuing the new CLI command or
setting the new pjproject.conf option.
Other options that can affect the logging are how you have the pjproject
log levels mapped to Asterisk log types in pjproject.conf and if you have
configured Asterisk to log the DEBUG type messages. Altering the
pjproject.conf level mapping shouldn't be necessary for most installations
as the default mapping is sensible. Configuring Asterisk to log the DEBUG
message type is standard practice for collecting debug information.
* Added CLI "pjproject set log level" command to dynamically adjust the
maximum pjproject log message level.
* Added CLI "pjproject show log level" command to see the currently set
maximum pjproject log message level.
* Added pjproject.conf startup section "log_level" option to set the
initial maximum pjproject log message level so all messages could be
captured from initialization.
* Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into
bundled pjproject. Pjproject will use the currently set run time log
level to determine if a log message is generated just like Asterisk
verbose and debug logging levels.
* In log_forwarder(), made always log enabled and mapped pjproject log
messages. DEBUG mapped log messages are no longer gated by the current
Asterisk debug logging level.
* Removed RAII_VAR() from res_pjproject.c:get_log_level().
ASTERISK-26630 #close
Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
The sending codec is switched to the receiving codec and then
is switched back to the best native codec on EVERY receiving RTP packets.
This is because after call of ast_channel_set_rawwriteformat there is call
of ast_set_write_format which calls set_format which sets rawwriteformat
to the best native format.
This patch adds a new function ast_set_write_format_path which set
specific write path on channel and uses this function to switch
the sending codec.
ASTERISK-26603 #close
Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d
Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992
that requested ability to add callerid into app_originate.
Comments in that issue suggested that it was better solved by
adding an option to gosub prior to originating the call. The
attached patch implements this much like app_dial with two
options one to gosub on the originating channel and one to gosub
on the newly created channel and behaves just like app_dial.
I have tested this patch by adding callerid info to the new
channel and also SIPAddHeader (to e.g. add header to force auto
answer) and confirmed it works. Have also tested both 'exten'
and 'app' versions of app_originate.
Opened by: dkerr
Patch by: dkerr
Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57
Previously, a TLS server socket would only be restarted upon sip reload if the
bind address had changed. This commit adds checking for changes to TLS
parameters like certificate, ciphers, etc. so they get picked up without
requiring a reload of the entire chan_sip module. This does not affect open
connections in any way, but new connections will use the new TLS parameters.
The changes also apply to HTTP and Manager.
ASTERISK-26604 #close
Change-Id: I169e86cefc6dcd627c915134015a6a1ab1aadbe6
Responding to authentication challenges leaks PJSIP memory pools.
The leak was introduced with a pjproject 2.5.5 API change.
https://trac.pjsip.org/repos/ticket/1929 changed the API usage of
pjsip_auth_clt_init() to require the new API pjsip_auth_clt_deinit() to
clean up cached authentication allocations that get allocated with
pjsip_auth_clt_reinit_req().
ASTERISK-26516 #close
Change-Id: I4473141b8c3961d0dc91c382beb3876b3efb45c8
fopencookie/funclose is a non-standard API and should not be used
in portable software. Additionally, the way FILE's fd is used in
non-blocking mode is undefined behaviour and cannot be relied on.
This introduces internal abstraction for io streams, that allows
implementing the desired virtualization of read/write operations
with necessary timeout handling.
ASTERISK-24515 #close
ASTERISK-24517 #close
Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85
During the development of Asterisk 14 the behavior of
the Command AMI action was altered such that the result
was returned on lines with a prefix of "Output: ". While
this was documented in the UPGRADE.txt file it is also
reasonable that this should bump the AMI version number.
ASTERISK-26556
Change-Id: Idf1bf01608e53f7bfdf43ddb4d0683e53f74ee42
In multi-party bridges, Asterisk currently supports two video modes:
* Follow the talker, in which the speaker with the most energy is shown
to all participants but the speaker, and the speaker sees the
previous video source
* Explicitly set video sources, in which all participants see a locked
video source
Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.
This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
Removes any explicit video source, and sets the video mode to talk
detection
ASTERISK-26595 #close
Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621
This works the same as for AMI manager variables. Set
"channelvars=foo,bar" in your ari.conf general section, and then the
channel variables "foo" and "bar" (along with their values), will
appear in every Stasis websocket channel event.
ASTERISK-26492 #close
patches:
ari_vars.diff submitted by Mark Michelson
Change-Id: I5609ba239259577c0948645df776d7f3bc864229
Libedit 3.1 is not build with unicode on as a default and so the
prototype for the el_gets callback changed from expecting a char buffer
to accepting a wchar buffer. If ast_el_read_char isn't changed,
the cli reads garbage from teh terminal.
Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and
updated ast_el_read_char to use the HAVE_ define to detemrine whether
to use char or wchar.
ASTERISK-26592 #close
Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a
Radcli is yet another RADIUS client library, generally compatible with
freeradius and radiusclient-ng.
This commit adds autoconf option for detecting it as well and changes
cdr_radius and cel_radius to use its header file in that case.
ASTERISK-26540 #close
Change-Id: I271f0715406334874865ffbce0b354b3a2ca148f
There are several places in Asterisk that have duplicated logic
for deferring important frames until later.
This commit adds a couple of API calls to facilitate this automatically.
ast_channel_start_defer_frames(): Future reads of deferrable frames on
this channel will be deferred until later.
ast_channel_stop_defer_frames(): Any frames that have been deferred get
requeued onto the channel.
ASTERISK-26343
Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641
Adds an identifier (with a getter and setter) to detect channels with
interleaved audio.
This is needed by the binaural bridge_softmix patch (ASTERISK-26292) and
was already discussed here:
http://lists.digium.com/pipermail/asterisk-dev/2016-October/075900.html
The identifier can be set during fmtp parsing (to be seen in the
res_format_attr_opus.c change).
ASTERISK-26292
Change-Id: I359801cc5f98c35671c48dabc81a7f4ee1183d63
The readdir_r function has been deprecated and should no longer be used. This
patch removes the readdir_r dependency (replaced it with readdir) and also moves
the directory search code to a more centralized spot (file.c)
Also removed a strict dependency on the dirent structure's d_type field as it
is not portable. The code now checks to see if the value is available. If so,
it tries to use it, but defaults back to using the stats function if necessary.
Lastly, for most implementations of readdir it *should* be thread-safe to make
concurrent calls to it as long as different directory streams are specified.
glibc falls into this category. However, since it is possible that there exist
some implementations that are not safe, locking has been added for those other
than glibc.
ASTERISK-26412
ASTERISK-26509 #close
Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba
Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK
(Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the
dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges.
Consequently, when the dynamic range is exhausted, this change utilizes payload
types in the range between 35 and 63 giving room for another 29 payload types.
ASTERISK-26311 #close
Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964
This patch adds three new CLI commands:
- ari show apps: list the registered ARI applications
- ari show app: show detailed information about an ARI application
- ari set debug: dump events being sent to an ARI application
Note that while these CLI commands live in the res_stasis module, we use
the 'ari' family for these commands. This was done as most users of
Asterisk aren't aware of the semantic differences between ARI and
res_stasis, and some 'ari' CLI commands already exist.
ASTERISK-26488 #close
Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5
PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD
define it to a constant. It is indeed not safe to assume there won't be
longer paths and Asterisk generally does err safely on such cases.
So even for HURD we'll just pretend PATH_MAX is 4096.
ASTERISK-25070 #close
Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3
Headers declare that memcpy does not accept NULL argument for the first
two parameters. Add a conditional block to prevent memcpy and ast_free
from running on vectors with NULL element array.
ASTERISK-26526 #close
Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71
It is only safe to run ast_register_cleanup callbacks when all modules
have been unloaded. Previously these callbacks were run during graceful
shutdown, making it possible to crash during shutdown.
ASTERISK-26513 #close
Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
ASTERISK-26423 #close
Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
ARI and AMI allow for an explicit channel ID to be specified
when originating channels. Unfortunately, there is nothing in
place to prevent someone from using the same ID for multiple
channels. Further complicating things, adding ID validation to channel
allocation makes it impossible for ARI to discern why channel allocation
failed, resulting in a vague error code being returned.
The fix for this is to institute a new method for channel errors to be
discerned. The method mirrors errno, in that when an error occurs, the
caller can consult the channel errno value to determine what the error
was. This initial iteration of the feature only introduces "unknown" and
"channel ID exists" errors. However, it's possible to add more errors as
needed.
ARI uses this feature to determine why channel allocation failed and can
return a 409 error during origination to show that a channel with the
given ID already exists.
ASTERISK-26421
Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06
Since the json library does not make the check function public we
recreate/copy the function in our interface module.
ASTERISK-26466
Reported by: Richard Mudgett
Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99
Adds libfftw3 to the build chain that is is going to be used for binaural
synthesis by bridge_softmix.
ASTERISK-26292
Change-Id: Iedc2f174e4ccb39ae5d9e698e339c6a17155867b
Small fix. It is necessary to double-check
the index that we just removed because there
is a new element.
ASTERISK-26453 #close
Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7
Fixed a memory leak. It removes only the first element.
Added a useful feature in vector.h to remove all items
under the CMP through a callback function / macro.
ASTERISK-26453 #close
Change-Id: I84508353463456d2495678f125738e20052da950
We use a lot res_calendar, we are very happy with that, especially
because you use libical, the almost alone opensource library that
supports really ical format with all types of recurrency.
Nevertheless, some features are missed for our business use cases.
This first patch adds a new option in calendar.conf:
fetch_again_at_reload. Be my guest for a better name.
If it's true, when you'll launch "module reload res_calendar.so",
Asterisk will download again the calendar.
The business use case is that we have a WebUI with a scheduler planner,
we know when the calendars are modified.
For now, we need to define 1 minute of timeout to have a chance that
our user doesn't wait too long between the modification and the real
test. But it generates a lot of useless HTTP traffic.
ASTERISK-26422 #close
Change-Id: I384b02ebfa42b142bbbd5b7221458c7f4dee7077
Added tests for bzip2, tar, patch, sed and nm to configure.ac.
Set DOWNLOAD_TO_STDOUT to a working command line regardless of
whether the download program is wget, curl or fetch.
Added a 'configure.m4' file to the third-party directory which takes
care of calling any third-party project setup. Had to move some
pjproject_bundled stuff up in configure.ac so it was called before
the third-party configure macro.
The pjproject tarball is now downloaded to the externals_cache_dir if
it was specified on the ./configure command line
Removed regeneration of the pjproject aconfigure file. It was only
needed for an old patch that no longer applies.
Converted the tests for symbols to explicit tests since we know that
they're now available in the bundled version. Saves a little time
during configure.
ASTERISK-26416 #close
Reported-by: Corey Farrell
Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b
(cherry picked from commit e6b0053d75)
(cherry picked from commit a0d02f3832)
Asterisk only supports mono audio at the moment.
This patch adds interleaved two-channel audio to Asterisk's channels.
ASTERISK-26292
Change-Id: I7a547cea0fd3c6d1e502709d9e7e39605035757a
* Compile __ast_assert_failed unconditionally.
* Use __ast_assert_failed to log messages from log_bad_ao2
* Remove calls to ast_assert(0) that happen after log_bad_ao2 was run.
Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751
This allows asterisk to compiled with LOW_MEMORY to load modules built
without LOW_MEMORY.
ASTERISK-26398 #close
Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d
Add Ogg/Opus playback support.
This uses libopusfile in order to be able to read .opus files and play
them back.
Writing/recording support is not present at this time.
ASTERISK-26409
Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955
(cherry picked from commit daee8bbd5209b4158bc1785eede845a26e6cbeaa)
Users upgrading from asterisk 13.5 to a later version and who use
realtime with peers that have mailboxes were experiencing runaway
situations that manifested as a continuous stream of taskprocessor
congestion errors, memory leaks and an unresponsive chan_sip.
A related issue was that setting rtcachefriends=no NEVER worked in
asterisk 13 (since the move to stasis). In 13.5 and earlier, when a
peer tried to register, all of the stasis threads would block and
chan_sip would again become unresponsive. After 13.5, the runaway
would happen.
There were a number of causes...
* mwi_event_cb was (indirectly) calling build_peer even though calls to
mwi_event_cb are often caused by build_peer.
* In an effort to prevent chan_sip from being unloaded while messages
were still in flight, destroy_mailboxes was calling
stasis_unsubscribe_and_join but in some cases waited forever for the
final message.
* add_peer_mailboxes wasn't properly marking the existing mailboxes
on a peer as "keep" so build_peer would always delete them all.
* add_peer_mwi_subs was unsubscribing existing mailbox subscriptions
then just creating them again.
All of this was causing a flood of subscribes and unsubscribes on
multiple threads all for the same peer and mailbox.
Fixes...
* add_peer_mailboxes now marks mailboxes correctly and build_peer only
deletes the ones that really are no longer needed by the peer.
* add_peer_mwi_subs now only adds subscriptions marked as "new" instead
of unsubscribing and resubscribing everything. It also adds the peer
object's address to the mailbox instead of its name to the subscription
userdata so mwi_event_cb doesn't have to call build_peer.
With these changes, with rtcachefriends=yes (the most common setting),
there are no leaks, locks, loops or crashes at shutdown.
rtcachefriends=no still causes leaks but at least it doesn't lock, loop
or crash. Since making rtcachefriends=no work wasnt in scope for this
issue, further work will have to be deferred to a separate patch.
Side fixes...
* The ast_lock_track structure had a member named "thread" which gdb
doesn't like since it conflicts with it's "thread" command. That
member was renamed to "thread_id".
ASTERISK-25468 #close
Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0
sd_notify() is used to notify systemd of changes to the status of the
process. This allows the systemd daemon to know when the process
finished loading (and thus only start another program after Asterisk has
finished loading).
To use this, use a systemd unit with 'Type=notify' for Asterisk.
This commit also adds the function ast_sd_notify(), a wrapper around
sd_notify that does nothing if not built with systemd support.
Also adds support for libsystemd detection in the configure script.
Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811
This implements the chan_sip legacy_useroption_parsing option but with a
better name.
* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.
ASTERISK-26316 #close
Reported by: Kevin Harwell
Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn
Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762
On heavy loaded system the TCP/TLS incoming calls could be
disconnected by pjproject while these calls are being
processed by asterisk which could use the session's memory pools.
If the session in the disconnected state then the session memory
pools were already freed, so we get segfault.
This patch adds a lifetime control on an INVITE session to pjproject.
The lifetime of the session is manipulated by calling
pjsip_inv_add_ref/pjsip_inv_dec_ref.
This patch uses these functions to inform pjproject that the
session is in use.
This patch adds check if the session state is not disconnected
and also checks if the memory pool is not NULL.
This patch also places tasks 'session_end' and 'session_end_completion'
into session's serializer to avoid race condition.
ASTERISK-26291 #close
Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7
Create an alternative to ast_sorcery_generic_alloc which uses astobj2
shared locking. Use this new method for the 'struct ast_sip_aor' allocator.
Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f
This allows standard ao2 functions to be used to release references to
an ast_named_lock. This change can cause less frequent locking of the
global named_locks container. The container is no longer locked when a
named_lock reference is being release except when this causes the
named_lock to be destroyed.
Change-Id: I644e39c6d83a153d71b3fae77ec05599d725e7e6
ast_channel_get_t38_state() calls ast_channel_queryoption() with
AST_OPTION_T38_STATE. If the passed in channel is a local channel then a
deadlock can happen if a channel lock is held when called.
* Made ast_channel_get_t38_state() callers not hold a channel lock before
calling.
* Update ast_channel_get_t38_state() doxygen to note that no channel locks
can be held when calling the function.
ASTERISK-26203 #close
Reported by: Etienne Lessard
ASTERISK-24822 #close
Reported by: David Brillert
ASTERISK-22732 #close
Reported by: Richard Mudgett
Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214
ASTERISK-25980 added the FAXMODE channel variable to res_fax.c.
Unfortunately, it also introduced a deadlock potential because
set_channel_variables() which sets FAXMODE can be called during a
masquerade. The ast_channel_get_t38_state() which gets the value used to
set FAXMODE cannot be called with the channel locked. As a result, local
channels can deadlock because of how they must acquire the locks necessary
to operate.
The intent of FAXMODE is for dialplan to know how a fax was transferred
after the fax completes. However, the previous patch sets FAXMODE to the
channel's current T.38 state AFTER the fax has completed and where T.38
may have already disconnected.
* Set FAXMODE based upon T.38 negotiations exchanged either with the fax
applications or the fax framehooks.
ASTERISK-26203
Reported by: Etienne Lessard
ASTERISK-24822
Reported by: David Brillert
ASTERISK-22732
Reported by: Richard Mudgett
Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1
MALLOC_DEBUG should not be used to check if debugging is actually
enabled, __AST_DEBUG_MALLOC should be used instead. MALLOC_DEBUG only
indicates that debugging is requested, __AST_DEBUG_MALLOC indicates it
is active.
Change-Id: I3ce9cdb6ec91b74ee1302941328462231be1ea53
Allocator functions that take file/line/func parameters are prefixed
with single-underscore when MALLOC_DEBUG is not defined,
double-underscore when it is defined. This change updates all
allocators that accept file/line/func to have the same prototype in
either ABI mode. The parameter order of __ast_vasprintf and
__ast_asprintf in utils.h have been changed to match that of astmm.h.
End-use allocator macro's have been removed from astmm.h and moved to an
unconditional part of utils.h.
Change-Id: I823bb6ce2b5675b3a4735948f10a3b420e9a023a
updated the uri handler to include the url prefix of the http server
this enables res_ari to add it to the uris when generating docs
Change-Id: I279335a2625261a8492206c37219698f42591c2e
(cherry picked from commit 6f448f32fe)
contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.
Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
The Exchanging Device and Mailbox States could not working
if the Entity ID (EID) is not set manually and can't be obtained
from ethernet interface.
This patch replaces debug message to warning
and addes missing description about option 'entityid' to
asterisk.conf.sample.
With this patch the asterisk also:
(1) decline loading the modules which won't work without EID:
res_corosync and res_pjsip_publish_asterisk.
(2) warn if EID is empty on loading next modules:
pbx_dundi, res_xmpp
Starting with v197 systemd/udev will automatically assign "predictable"
names for all local Ethernet interfaces.
This patch also addes some new ethernet prefixes "eno" and "ens".
ASTERISK-26164 #close
Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6
A patch made to the master branch (Now the 14 branch) inadvertently made
libsrtp a required dependency in order to compile Asterisk. Rather than
create dummy defines to substitute for the defines supplied by libsrtp
when libsrtp is not available, most of the code in sdp_srtp.c is moved
into res_srtp.c. This gets more code out of Asterisk's core that isn't
used when SRTP is not available. This also makes another inadvertent
required dependency on libsrtp by Asterisk's core unlikely.
ASTERISK-26253 #close
Reported by: Ben Merrills
Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7
If both channels which should be masqueraded
are in the same serializer:
1st channel will be locked waiting condition 'complete'
2nd channel will be locked waiting condition 'suspended'
On heavy load system a chance that both channels will be in
the same serializer 'pjsip/distibutor' is very high.
To reproduce compile res_pjsip/pjsip_distributor.c with
DISTRIBUTOR_POOL_SIZE=1
Steps to reproduce:
1. Party A calls Party B (bridged call 'AB')
2. Party B places Party A on hold
3. Party B calls Voicemail app (non-bridged call 'BV')
4. Party B attended transfers Party A to voicemail using REFER.
5. When asterisk masquerades calls 'AB' and 'BV',
a deadlock is happened.
This patch adds a suspension indicator to the taskprocessor.
When a session suspends/unsuspends the serializer
it sets the indicator to the appropriate state.
The session checks the suspension indicator before
suspend the serializer.
ASTERISK-26145 #close
Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b
libunbound at version 1.4.20 (which CentOS still uses) declared all
of their string function parameters as as 'char *'. 1.4.21 changed
them all to 'const char *'. Thankfully 1.4.21 also introduced the
UNBOUND_VERSION_MAJOR define so configure now checks for that and
sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks
that and casts away the 'const' if it's not set.
Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and
Fedora24 (1.5.4). There are a few failing tests to be addressed though.
ASTERISK-26283 #close
Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148
Modules must define AST_MODULE_SELF_SYM to be used as the name of a
generated function. This produces a friendly error when it's not
defined.
ASTERISK-26278 #close
Change-Id: Ib9d35a08104529c516d636771365e02c6e77a45b
George Joseph
Lorenzo Miniero
Kevin Harwell
Aaron An
zuul
Jonathan R. Rose
Richard Mudgett
Joshua Colp
Guido Falsi
Alexei Gradinari
David Kerr
Michael Kuron
Mark Michelson
Timo Teräs
Matt Jordan
Sebastien Duthil
Tzafrir Cohen
frahaase
Alexander Traud
Corey Farrell
Badalyan Vyacheslav
Ludovic Gasc (GMLudo)
Torrey Searle