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To be consistent with sdp implementation.
Change-Id: I714e300939b4188f58ca66ce9d1e84b287009500
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This patch fixes 2 original issues and more that those 2 exposed.
* When we send a NOTIFY, and the client either doesn't respond or
responds with a non OK, pjproject only calls our
pubsub_on_evsub_state callback, no others. Since
pubsub_on_evsub_state (which does the sub_tree cleanup) does not
expect to be called back without the other callbacks being called
first, it just returns leaving the sub_tree orphaned. Now
pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE
which is what pjproject will set to tell us that it was the
transaction that timed out or failed and not the subscription
itself timing our or being terminated by the client. If is
TSX_STATE, pubsub_on_evsub_state now does the proper cleanup
regardless of the state of the subscription.
* When a client renews a subscription, we don't update the
persisted subscription with the new expires timestamp. This causes
subscription_persistence_recreate to prune the subscription if/when
asterisk restarts. Now, pubsub_on_rx_refresh calls
subscription_persistence_update to apply the new expires timestamp.
This exposed other issues however...
* When creating a dialog from rdata (which sub_persistence_recreate
does from the packet buffer) there must NOT be a tag on the To
header (which there will be when a client refreshes a
subscription). If there is one, pjsip_dlg_create_uas will fail.
To address this, subscription_persistence_update now accepts a flag
that indicates that the original packet buffer must not be updated.
New subscribes don't set the flag and renews do. This makes sure
that when the rdata is recreated on asterisk startup, it's done
from the original subscribe packet which won't have the tag on To.
* When creating a dialog from rdata, we were setting the dialog's
remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq.
When the client tried to resubscribe after a restart with the
correct cseq, we'd reject the request with an Invalid CSeq error.
* The acts of creating a dialog and evsub by themselves when
recreating a subscription does NOT restart pjproject's subscription
timer. The result was that even if we did correctly recreate the
subscription, we never removed it if the client happened to go away
or send a non-OK response to a NOTIFY. However, there is no
pjproject function exposed to just set the timer on an evsub that
wasn't created by an incoming subscribe request. To address this,
we create our own timer using ast_sip_schedule_task. This timer is
used only for re-establishing subscriptions after a restart.
An earlier approach was to add support for setting pjproject's
timer (via a pjproject patch) and while that patch is still included
here, we don't use that call at the moment.
While addressing these issues, additional debugging was added and
some existing messages made more useful. A few formatting changes
were also made to 'pjsip show scheduled tasks' to make displaying
the subscription timers a little more friendly.
ASTERISK-26696
ASTERISK-26756
Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e
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Adds topology set and get to channel.
ASTERISK-26790
Change-Id: Ic379ea82a9486fc79dbd8c4d95c29fa3b46424f4
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This change adds the media stream topology definition and API for
accessing and using it.
Some refactoring of the stream was also done.
ASTERISK-26786
Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568
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In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c49. This patch
restores that behavior.
This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.
ASTERISK-26115 #close
Reported by: Nasir Iqbal
Change-Id: I8afbd06725e99610e02adb529137d4800c05345d
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This change adds the media stream definition and API for
accessing and using it. Unit tests have also been written
which exercise aspects of the API.
ASTERISK-26773
Change-Id: I3dbe54065b55aaa51f467e1a3bafd67fb48cac87
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A dialplan intercept routine is equivalent to an interrupt routine. As
such, the routine must be done quickly and you do not have access to the
media stream. These restrictions are necessary because the media stream
is the responsibility of some other code and interfering with or delaying
that processing is bad. A possible future dialplan processing
architecture change may allow the interception routine to run in a
different thread from the main thread handling the media and remove the
execution time restriction.
* Made res_agi.c:run_agi() running an AGI in an interception routine run
in DeadAGI mode. No touchy channel frames.
ASTERISK-25951
ASTERISK-26343
ASTERISK-26716
Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43
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There are several issues with deferring frames that are caused by the
refactoring.
1) The code deferring frames mishandles adding a deferred frame to the
deferred queue. As a result the deferred queue can only be one frame
long.
2) Deferrable frames can come directly from the channel driver as well as
the read queue. These frames need to be added to the deferred queue.
3) Whoever is deferring frames is really only doing the __ast_read() to
collect deferred frames and doesn't care about the returned frames except
to detect a hangup event. When frame deferral is completed we must make
the normal frame processing see the hangup as a frame anyway. As such,
there is no need to have varying hangup frame deferral methods. We also
need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real.
That fake hangup is to cause the PBX thread to break out of loops to go
execute a new dialplan location.
4) To properly deal with deferrable frames from the channel driver as
pointed out by (2) above, means that it is possible to process a dialplan
interception routine while frames are deferred because of the
AST_CONTROL_READ_ACTION control frame. Deferring frames is not
implemented as a re-entrant operation so you could have the unsupported
case of two sections of code thinking they have control of the media
stream.
A worse problem is because of the bad implementation of the AMI PlayDTMF
action. It can cause two threads to be deferring frames on the same
channel at the same time. (ASTERISK_25940)
* Rather than fix all these problems simply revert the API refactoring as
there is going to be only autoservice and safe_sleep deferring frames
anyway.
ASTERISK-26343
ASTERISK-26716 #close
Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496
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mod_format.h: Note ast_filestream.fr holds a format ref.
translate.h: Note ast_trans_pvt.f holds a format ref.
Change-Id: I86bda354d725207b41e08920355d7c31b2d7f749
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The mechanism used for detecting the maximum log level compiled into the
linked pjproject did not work. The API call simply stores the requested
level into an integer and does no range checking. Asterisk was assuming
that there was range checking and limited the new value to the allowable
range. To get the actual maximum log level compiled into the linked
pjproject we need to get and save off the initial set log level from
pjproject. This is the maximum log level supported.
* Get and save off the initial log level setting before altering it to the
desired level on startup. This has to be done by a macro rather than
calling a core function to avoid incorrectly linking pjproject.
* Split the initial log level warning messages to warn if the linked
pjproject cannot support the requested startup level and if it is too low
to get the pjproject buildopts for "pjproject show buildopts".
* Adjust the CLI "pjproject set log level" to check the saved max log
level and to generate normal output messages instead of a warning message.
ASTERISK-26743 #close
Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4
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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 1d890874f39a5a81b20da44358143ed9b54ab0fe)
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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
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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
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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
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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
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ASTERISK-25083
Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2
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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
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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
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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
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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
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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
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ast_format"
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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
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Based on bridge video AMI event changes, bump the minor version of AMI.
Change-Id: Idf84507354170400813cda780906c94c9f1b60b4
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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
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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
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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
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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
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ASTERISK-26343
Change-Id: I06dbf7366e26028251964143454a77d017bb61c8
(cherry picked from commit 0be46aaf6b8b9eb5b0160ec591cdc2c6e1802a6d)
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