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Not easy to reproduce, but we have noticed deadlocks when unloading a module
while dialplan is handling a request.
The deadlock is between :
1) Dialplan execution: pbx_extension_helper() first taking conlock,
then pbx_findapp() [when called] asking for lock on apps list.
2) Application unregistration: ast_unregister_application() first taking lock
on apps list, then unreference_cached_app() [when called] asking for conlock.
As a protection, I suggest to modify ast_unregister_application(), so that it
anticipates the need of conlock, before taking the lock on apps list.
The side effect is a longer unavailability of conlock when unregistering an
application.
ASTERISK-27041
Change-Id: I0db0f1eb320da6a5758cce3a47d765be1face8e2
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* changes:
SDP: Set the remote c= line in RTP instance.
SDP: Add t= line in sdp_create_from_state()
stream: Ignore declined streams for some topology calls.
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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
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Closing IMAP connection on module reload or unload.
ASTERISK-24052 #close
Change-Id: I2a40182aa9ef249fa6865d33570430e9ada68525
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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
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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
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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
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Change-Id: I82dc75c63c48904e9e5a49e2205dcc06e88487e4
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* Pulled finding the rtcp-mux attribute flag out of the ICE candidate for
loop. Also ordered the RTCP ICE candidate skip test to fail earlier.
Change-Id: I8905d9c68563027a46cd3ae14dbcc27e9c814809
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Change-Id: I23b646392082deab65bedeb19b12dcbcb9216d0c
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Change-Id: If07e3c716a2e3ff85ae905c17572ea6ec3cdc1f9
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Change-Id: I4060391328a893101ed87d0d9bacbbab4fd8b141
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* Made ast_format_cap_from_stream_topology() not include any formats from
declined streams.
* Made ast_stream_topology_get_first_stream_by_type() ignore declined
streams to return the first active stream of the type.
* Updated unit tests to check these changes have the expected effect.
Change-Id: Iabbc6a3e8edf263a25fd3056c3c614407c7897df
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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
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Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1
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Closing IMAP connection on MWI unsubscribe.
ASTERISK-24052 #close
Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd
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destroy_subscription was attempting to get the id of the
subscription tree's endpoint after we'd already called ao2_cleanup
on it causing a segfault.
Moved the cleanup until after the debug statement and since
endpoint could also be NULL at this point, check for that as well.
ASTERISK-27057 #close
Reported-by: Ryan Smith
Change-Id: Ice0a7727f560cf204d870a774c6df71e159b1678
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There was a typo introduced in commit 776ffd77 which was preventing
the transport's external media address from being used.
ASTERISK-27024 #close
Reported-by: Christopher van de Sande
patches:
patch.diff submitted by Florian Floimair (license 6892)
Change-Id: I7ec617171eaa2d86d2680b00cf37d5088adafc27
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It looks like there was a copy/paste error in ast_rtp_change_source
where if there was a rtcp srtp instance, instead of updating its
ssrc we were updating the srtp instance ssrc twice.
ASTERISK-27022 #close
Reported-by: Michael Walton
Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095
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This change adds a deferred queue to bridging. If a bridge
technology determines that a frame can not be written and
should be deferred it can indicate back to bridging to do so.
Bridging will then requeue any deferred frames upon a new
channel joining the bridge.
This change has been leveraged for T.38 request negotiate
control frames. Without the deferred queue there is a race
condition between the bridge receiving the T.38 request
negotiate and the second channel joining and being in the
bridge. If the channel is not yet in the bridge then the T.38
negotiation fails.
A unit test has also been added that confirms that a T.38
request negotiate control frame is deferred when no other
channel is in the bridge and that it is requeued when a new
channel joins the bridge.
ASTERISK-26923
Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415
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When doing an attended transfer it's possible for the transferer, after
receiving an accepted response from Asterisk, to send a BYE to Asterisk,
which can then be processed before Asterisk has time to start and/or
complete the transfer process. This of course causes the transfer to not
complete successfully, thus dropping the call.
This patch makes it so any BYEs received from the transferer, after the REFER,
that initiate a session end are deferred until the transfer is complete. This
allows the channel that would have otherwise been hung up by Asterisk to
remain available throughout the transfer process.
ASTERISK-27053 #close
Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a
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We now mirror the pjproject tarball and md5 at
https://github.com/asterisk/third-party/tree/master/pjproject
To improve download reliability, we now get the tarball from
our mirror instead of from pjsip.org.
ASTERISK-27052 #close
Reported-by: 'alex'
Change-Id: I60236587a8935bfa71fcc391f4e2ecb31918c08a
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If sending unsolicited mwi to all endpoints on startup is disabled
(mwi_disable_initial_unsolicited=yes) do not need to create subscriptions.
If there are many (thousands) realtime endpoints configured with unsolicited mwi
and Vociemail Storage configured as ODBC or IMAP there will be huge number of
DB/IMAP requests on startup.
ASTERISK-26230 #close
Change-Id: I50ae909639e3ee298b931a54def4b2b9e0fb86c5
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Reported by Sylvain Boily via asterisk-dev mailing list.
Change-Id: Idc7623f335aea3e144dd369ba383b9a757480a9d
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Added check for NULL return value when calling
ast_sorcery_retrieve_by_id in function get_write_timeout
ASTERISK-27046
Change-Id: I9357717278da631c3a1cb502c412693929b0cb41
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Add some #if defined checks which allow building against LibreSSL.
These patchess come from OpenBSD ports:
https://cvsweb.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/patches/
ASTERISK-27043 #close
Reported by: OpenBSD ports
Change-Id: I2f6c08a5840b85ad4d2b75370b947ddde7a9a572
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Some systems (like macOS) require BIND_8_COMPAT to be defined so that
the nameser libraries are, well, BIND8 compatible.
Change-Id: If79fc27a64f90de1835b5aa3aadfa9be22bd16b0
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FreeBSD does not include a crypt.h include file. Definitions for
crypt() and crypt_r() are in unistd.h
ASTERISK-27042 #close
Change-Id: Ib307ee5e384870c6af50efa89fb73722dd0c3a7e
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The chan_pjsip module uses a calculation approach for
determining device state. This means that in situations
where we would expect device state to change we need to
tell the core to query. A scenario that was missed is
when early media was signaled.
This change adds the notification for the core to
query device state when we are told that early media
is being provided.
ASTERISK-27039
Change-Id: Iafebfd152894966344ff2e950a3cee9f59a3eb6f
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