The primary focus of this patch is adding a missing call to
ast_odbc_release_obj(), but is also a general cleanup of the ODBC
related code in app_voicemail.
ASTERISK-27093 #close
Change-Id: I8e285142eaeb3146b4287a928276b70db76c902b
Clear channel flag AST_FLAG_END_DTMF_ONLY in ast_waitfordigit_full when
ast_read returns NULL.
ASTERISK-27100 #close
Change-Id: Id3039e9a4e74e0cb359f636c9fd0c9740ebf7d9d
When a SIP message comes in on a transport, pjproject obtains the lock on
the transport and pulls the data out of the socket. Unlike UDP, the TCP
transport does not allow concurrent access. Without concurrency the
transport lock is not released when the transport's message complete
callback is called. The processing continues and eventually Asterisk
starts processing the SIP message. The first thing Asterisk tries to do
is determine the associated dialog of the message to determine the
associated serializer. To get the associated serializer safely requires
us to get the dialog lock.
To send a request or response message for a dialog, pjproject obtains the
dialog lock and then obtains the transport lock. Deadlock can result
because of the opposite order the locks are obtained.
* Fix the deadlock by obtaining the serializer associated with the dialog
another way that doesn't involve obtaining the dialog lock. In this case,
we use an ao2 container to hold the associated endpoint and serializer.
The new locks are held a brief time and won't overlap other existing lock
times.
ASTERISK-27090 #close
Change-Id: I9ed63f4da9649e9db6ed4be29c360968917a89bd
The OBJ_SEARCH_xxx defines should not be used as if they were individual
bits. They represent a multi-bit enumeration value field.
Change-Id: I32abc9a475396dab02402a7014357dd94284e17b
There wasn't any good way to pass options like --host or --build
down to the pjproject configure which makes cross-compiling difficult.
* Added a new PJPROJECT_CONFIGURE_OPTS environment variable which
can be used to pass arbitrary options to pjproject configure.
* Automatically set the pjproject configure --host and --build
options to match those supplied for the asterisk configure.
ASTERISK-27097 #close
Reported-by: Kinsey Moore
Change-Id: I5fa776e110262851173002a26ffe1172e4c35b2e
When connected_line_method is "invite", we're supposed to determine
if the client can support UPDATE and if it can, send UPDATE instead
of INVITE to avoid the SDP renegotiation. Not only was pjproject
not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing
that invite_tsx wasn't NULL which isn't always the case.
* Updated chan_pjsip/update_connected_line_information to drop the
requirement that invite_tsx isn't NULL.
* Submitted patch to pjproject sip_inv.c that sets the
PJSIP_INV_SUPPORT_UPDATE flag correctly.
* Updated pjsip.conf.sample to clarify what happens when "invite"
is specified.
ASTERISK-27095
Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560
The fix for ASTERISK-25665 introduced a regression.
The return value of queue_exec used to be 0 in case of leavewhenempty
but it was changed to -1 (returned from wait_our_turn and passed
transparently by queue_exec), thus leading to hangup instead of returning
back to dialplan.
This commit resets the value back to 0 in this case, restoring
original behavior.
ASTERISK-27065 #close
Reported by: Marek Cervenka
Change-Id: Id9c83b75aeda463250155e88c5004be52bbca5ac
When re-inviting to add more streams it is possible for
the role of existing ICE sessions to be changed to the
incorrect value. This results in subsequent refreshes
within the sessions getting a role conflict and the ICE
session breaking down. This change only sets the role to
be the new value if an ICE renegotiation is actually
going to happen, otherwise the existing role is preserved.
As well if we encounter a situation where a unidirectional
ICE negotiation happens and the other side does not send us
candidates we will not store any information for sending
traffic, even though we know where they are reachable. This
change fixes this by using the source of the ICE traffic
itself as the target if no candidates are known and we
receive some ICE traffic.
ASTERISK-27088
Change-Id: I71228181e358917fcefc3100fad21b2fc02a59a9
There have been reports of deadlocks caused by an attempt to send a frame
to a channel's rtp instance after the channel has left the native bridge
and been destroyed. This patch effectively causes the bridge channel to
keep a reference to the glue and both the audio and video rtp instances
so what gets started will get stopped.
ASTERISK-26978 #close
Reported-by: Ross Beer
Change-Id: I9e1ac49fa4af68d64826ccccd152593cf8cdb21a
The T38 sdp callback incorrectly has a side effect of incrementing
the media_count. This can lead to core dumps.
Change-Id: I7bb2f4987de4046ec52cfc34e5ea0662dae32af8
The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated. This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.
ASTERISK-27066 #close
Change-Id: Id185b11e84afd9191a2f269e8443019047765e91
arm the t38 webhook always, so we can correctly reject a
T38 negotiation request when t38 is disabled on a channel
Change-Id: Ib1ffe35aee145d4e0fe61dd012580be11aae079d
Do not need to unsubscribe/subscribe on creating the ednpoint's contact.
The modified function create_mwi_subscriptions_for_endpoint adds
the subscription only if it does not exist.
The subscriptions aren't added for active contacts
which are retrieved on startup from realtime
if mwi_disable_initial_unsolicited=yes.
Because the mwi_contact_added is not called.
So the subscriptions also should be created on updating contact.
ASTERISK-26230 #close
Change-Id: I47e265af9296ca09aa42a316fdacac104148cee4
In an earlier version of Asterisk a local channel [un]lock all functions were
added in order to keep a crash from occurring when a channel hung up too early
during an attended transfer. Unfortunately, when a transfer failure occurs and
depending on the timing, the local channels sometime do not get properly
unlocked and deref'ed after being locked and ref'ed. This happens because the
underlying local channel structure gets NULLed out before unlocking.
This patch reworks those [un]lock functions and makes sure the values that get
locked and ref'ed later get unlocked and deref'ed.
ASTERISK-27074 #close
Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09
If an attended transfer failed it was possible for some of the channels
involved to get "stuck" because Asterisk was not hanging up the transfer target.
This patch ensures Asterisk hangs up the transfer target when an attended
transfer failure occurs.
ASTERISK-27075 #close
Change-Id: I98a6ecd92d3461ab98c36f0d9451d23adaf3e5f9
A new global option "imap_poll_logout" was added to specify whether need to
disconnect from the IMAP server after polling of mailboxes.
ASTERISK-27068 #close
Closing IMAP connection after loading mailbox from voicemail.conf
ASTERISK-24052 #close
Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a
If the endpoint's last contact is deleted unsolicited MWI has to be
unsubscribed.
ASTERISK-27051 #close
Change-Id: I33e174e0b9dba0998927d16d6d100fda5c7254e0
When a stasis channel is stolen by another app, the control
structure is unreffed but never unlinked from the app_controls
container. This causes the channel reference to leak.
Added OBJ_UNLINK to the callback in channel_stolen_cb.
Also added some additional channel lifecycle debug messages to
channel.c.
ASTERISK-27059 #close
Repoorted-by: George Joseph
Change-Id: Ib820936cd49453f20156971785e7f4f182c56e14
This option was added to control whether to notify dialog-info state
'early' or 'confirmed' on Ringing when already INUSE.
The value "yes" is useful for some SIP phones (Cisco SPA)
to be able to indicate and pick up ringing devices.
ASTERISK-26919 #close
Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711
In Corosync 2.x libraries were changed to use LibQB IPC.
Sadly LibQB IPC doesn't support copy-free access to received buffer, so
Corosync libraries were rewritten to use stack as buffer. Mostly the
needed stack size is quite small, but for all *_dispatch functions, 1MiB
is needed.
Asterisk function ast_pthread_create_background set stack size for new
thread to much smaller AST_BACKGROUND_STACKSIZE (~500KiB).
This results in Asterisk crash when running with Corosync 2.x.
Patch solves this issue by creating it's own version of
ast_pthread_create_background which sets stack size to much higher value
(actually it's AST_BACKGROUND_STACKSIZE + 3MiB).
Another problem may appear when "corosync show members" netconsole
command is executed. It is also executed in thread and also has only
500KiB stack size. Sadly it calls corosync_cfg_get_node_addrs which
again needs at least 1MiB stack.
Solution is to use HAVE_COROSYNC_CFG_STATE_TRACK as a discriminator
between Corosync 1.x and 2.x. If 1.x is found, nothing changes. If 2.x
is found, NodeID is displayed instead of IP address.
ASTERISK-25370 #close
Reported by: mdu113
Change-Id: Id95b0d21ab6e708e7d74ad8786c587211676fa08
The recent change to make the use of LOAD_DECLINE more consistent
caused res_ari to unload itself before declining if the ari.conf
file wasn't found. The ari stubs though still tried to use the
configuration resulting in segfaults.
This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests
to see if res_ari is actually loaded and causes the stubs to also
decline if it isn't. The macro was then added to the mustache
template's "load_module" function.
ASTERISK-27026 #close
Reported-by: Ronald Raikes
Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d
The construction of the returned string assumed incorrectly that the
supplied buffer would always be initialized as an empty string. If it is
not an empty string we could overrun the supplied buffer by the length of
the non-empty buffer string plus one. It is also theoreticaly possible
for the supplied buffer to be overrun by a string terminator during a read
operation even if the supplied buffer is an empty string.
* Fix the assumption that the supplied buffer would already be an empty
string. The buffer is not guaranteed to contain an empty string by all
possible callers.
* Fix string terminator buffer overrun potential.
Change-Id: If6a0806806527678c8554b1dcb34fd7808aa95c9
The ast_channel_suppress function wrongly decremented the
reference count of the underlying structure used to keep
track of what should be suppressed on a channel if the
function was called multiple times on the same channel.
This change cleans up the reference counting a bit so
this no longer occurs.
ASTERISK-27016
Change-Id: I2eed4077cb4916e6626f9f120b63b963acc5c136
