The service is triggered by R5, i.e. flash hook + 5.

Grzegorz Sluja authored 2 years ago and Yalu Zhang committed 2 years ago

SIPSessionID is from Session-ID header field either in INVITE for incoming calls or in
200 OK for outgoing calls.

Grzegorz Sluja authored 2 years ago and Yalu Zhang committed 2 years ago

- Call waiting is enabled/disabled now per feature_set. Each line has the feature_set defined and each
  provider (pjsip endpoint) has line selected. From now on call waiting status can be defined in uci
  config and changed by feature code, as a result corresponding feature set or endpoint cw status will
  be changed
- Rename some functions and variables which had misleading names
- Add 5s beep timer indicating incoming call waiting
- Fix 20s timeout when there is already another call in progress
- Support call waiting/3 way call for DECT
- Implement "exceed call count" checking for line/extension/all

Adds a new configuration option, stir_shaken_profile, in pjsip.conf that
can be specified on a per endpoint basis. This option will reference a
stir_shaken_profile that can be configured in stir_shaken.conf. The type
of this option must be 'profile'. The stir_shaken option can be
specified on this object with the same values as before (attest, verify,
on), but it cannot be off since having the profile itself implies wanting
STIR/SHAKEN support. You can also specify an ACL from acl.conf (along
with permit and deny lines in the object itself) that will be used to
limit what interfaces Asterisk will attempt to retrieve information from
when reading the Identity header.

ASTERISK-29476

Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406

Change-Id: Ic15cfca9e68efd06a1b12ab2335d52a5890e7170

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

There was a possibility that during attended call transfer, deadlock
happened in locking bridge_channel and channel.
bridge_channel_queue_deferred_frames() was waiting for brcm_indicate()
to unlock the channel, while brcm_indicate() which called
ast_bridge_channel_queue_frame() was waiting for
bridge_channel_queue_deferred_frames() to unlock the bridge.

Creating new macro to lock both channel and bridge_channel at the same
time and sched_yield() -> retry if failed and use it in
bridge_channel_queue_deferred_frames() avoiding the deadlock.

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

This commit includes 3 fixes:
- change place of get/set rtp statistics to have them properly updated in cdr
for both incoming and outgoing call
- do not create additional "hangup" call log record
- save rtp stats counters from endptmngr into brcm_subchannel instead of
one global structure to support concurrent calls simultaneously

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Yalu Zhang authored 3 years ago and Sukru Senli committed 3 years ago

The bug was caused by a synchronous ubus API ubus_invoke(). The asynchronous version
of API ubus_invoke_aync() shall be used instead.

Also remove some unused code and stop ubus thread when the chan_brcm module is unloaded.

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

There is an issue with copying RTP statistics from snapshot to cdr hence
call log does not contains RTP stats for incoming calls - for outgoing
call it works fine.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

MWI feature can be enabled/disabled by the 'mwi_enabled' configuration
parameter in asterisk config. When the NOTIFY message is received by
pjsip with new message the ubus event is sent in asterisk.mwi path.
Apart from that 'mwi_dialtone_state' can be configured for specific
endpoint (sip account) - it will be used as audiable indication when new
message is waiting for the mailbox connected with the endpoint.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Change implementation of adding mediasec headers in REGISTER, INVITE,
REINVITE, OPTIONS events to be dynamically configured based on the
response from the Sip Server.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Yalu Zhang authored 4 years ago and Sukru Senli committed 3 years ago

It is caused that PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP is not defined on Broadcom ARM platform.
But recursive mutex is mandatory for Asterisk to work properly. Otherwise there will be dead lock
in some modules, e.g. loader.c.

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Added functions to open, close, and apply XML Stylesheets
to XML documents.  Although the presence of libxslt was already
being checked by configure, it was only happening if xmldoc was
enabled.  Now it's checked regardless.

Added ability to parse a string consisting of comma separated
name/value pairs into an ast_variable list.  The reverse of
ast_variable_list_join().

Change-Id: I1e1d149be22165a1fb8e88e2903a36bba1a6cf2e

Added:

Replace a variable in a list:
int ast_variable_list_replace_variable(struct ast_variable **head,
    struct ast_variable *old, struct ast_variable *new);
Added test as well.

Create a "name=value" string from a variable list:
'name1="val1",name2="val2"', etc.
struct ast_str *ast_variable_list_join(
    const struct ast_variable *head, const char *item_separator,
    const char *name_value_separator, const char *quote_char,
    struct ast_str **str);
Added test as well.

Allow the name of an XML element to be changed.
void ast_xml_set_name(struct ast_xml_node *node, const char *name);

Change-Id: I330a5f63dc0c218e0d8dfc0745948d2812141ccb

Adds the lastcontext and lastexten channel fields to allow users
to access previous dialplan execution locations.

ASTERISK-29840 #close

Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307

Currently, each module that uses libcurl duplicates the standard
Asterisk curl user agent.

This adds a global macro for the Asterisk user agent used for
curl requests to eliminate this duplication.

ASTERISK-29861 #close

Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d

Refactors temp file utility function into file.c.

ASTERISK-29809 #close

Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b

Whereas BLFs allow to show a display name for each RLS entry,
the asterisk provides only the extension now.
This is not end user friendly.

This commit adds a new resource_list option, resource_display_name,
to indicate whether display name of resource or the resource name being
provided for RLS entries.
If this option is enabled, the Display Name will be provided.
This option is disabled by default to remain the previous behavior.
If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
will be set as the Display Name.
The 'message-summary' is not supported yet.

ASTERISK-29891 #close

Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681

Adds a new option, defaultenabled, to the CDR core to
control whether or not CDR is enabled on a newly created
channel. This allows CDR to be disabled by default on
new channels and require the user to explicitly enable
CDR if desired. Existing behavior remains unchanged.

ASTERISK-29808 #close

Change-Id: Ibb78c11974bda229bbb7004b64761980e0b2c6d1

Previously there was no way to specify a connection timeout when
attempting to connect a websocket client to a server. This patch
makes it possible to now do such.

Change-Id: I5812f6f28d3d13adbc246517f87af177fa20ee9d

autoconfigh.h.in was missed in the original review for this
issue. Additionally it looks like I have newer pkg-config autoconf
macros on my development machine.

ASTERISK-29817

Change-Id: I3c85a4de82c5d7d6e0e23dad4c33bb650a86a57b

sched: Avoid a double deref when AST_SCHED_DEL_UNREF is called on an
executing call-back. This is done by adding a new variable 'rescheduled'
to the struct sched which is set in ast_sched_runq and checked in
ast_sched_del_nonrunning. ast_sched_del_nonrunning is a replacement for
now deprecated ast_sched_del which returns a new possible value -2
if called on an executing call-back with rescheduled set. ast_sched_del
is modified to call ast_sched_del_nonrunning to maintain existing code.
AST_SCHED_DEL_UNREF is also updated to look for the -2 in which case it
will not throw a warning or invoke refcall.
test_sched: Add a new unit test sched_test_freebird that will check the
reference count in the resolved scenario.

ASTERISK-29698

Change-Id: Icfb16b3acbc29cf5b4cef74183f7531caaefe21d

Remove the HMAC declarations from the includes.  They are
not implemented nor used anywhere, and their presence breaks the build
on NetBSD that delivers an incompatible hmac() function in <stdlib.h>.

ASTERISK-29818

Change-Id: I0c4b88645e30174b1b63846a6b328625b69c2ea7

Fixes CNG description from "noice" to "noise".

ASTERISK-29855 #close

Change-Id: Ie7cbbd7d72b426693df7447384ff8700318cd36d

Added two new functions to assist checking media types...

* ast_sip_are_media_types_equal compares two pjsip_media_types.
* ast_sip_is_media_type_in tests if one media type is in a list
  of others.

Added static definitions for commonly used media types to
res_pjsip.h.

Changed several modules to use the new functions and static
definitions.

ASTERISK_29813
(not ready to close)

Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9

There are times when you need to troubleshoot issues with bundled
pjproject or add new features that need to be pushed upstream
but...

* The source directory created by extracting the pjproject tarball
  is not scanned for code changes so you have to keep forcing
  rebuilds.
* The source directory isn't a git repo so you can't easily create
  patches, do git bisects, etc.
* Accidentally doing a make distclean will ruin your day by wiping
  out the source directory, and your changes.
* etc.

This commit makes that easier.
See third-party/pjproject/README-hacking.md for the details.

ASTERISK-29824

Change-Id: Idb1251040affdab31d27cd272dda68676da9b268

Adds an AMI event for a wink frame.

ASTERISK-29830 #close

Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56

The current TCP client connect code, blocks and does not handle EINTR
error case.

This patch makes the client socket non-blocking while connecting,
ensures a connect does not immediately fail due to EINTR "errors",
and adds a connect timeout option.

The original client start call sets the new timeout option to
"infinite", thus making sure old, orginal behavior is retained.

ASTERISK-29746 #close

Change-Id: I907571843a83e43c0742b95a64785f4411f02671