Chris-Savinovich [Sat, 19 Jan 2019 21:55:20 +0000 (15:55 -0600)]
Test_cel: Fails when DONT_OPTIMIZE is off
A bug in GCC causes TEST_CEL to return failure under the following
conditions:
1. TEST_FRAMEWORK on
2. DONT_OPTIMIZE off
3. Fedora and Ubuntu
4. GCC 8.2.1
5. Test name: test_cel_dial_pickup
6. There must exist a certain combination of multithreading.
The bug affects arithmetic calculations when the optimization level
is bigger than O1 and the -fpartial-inline flag is on. Provided these
conditions, function ast_str_to_lower() fails to convert to lower case
due to said function being of type force_inline. The solution is to
remove the "force_inline" type declaration from function ast_str_to_lower()
Sean Bright [Fri, 4 Jan 2019 23:14:45 +0000 (18:14 -0500)]
res_pjsip_transport_websocket: Don't assert on 0 length payloads
When --enable-dev-mode is used, pjsip_tpmgr_receive_packet() will assert
if passed a payload length of 0, so treat empty frames as if we didn't
receive them.
mohitdhiman [Mon, 7 Jan 2019 18:04:43 +0000 (23:34 +0530)]
stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure.
During Bridging of two channels if masquerade operation is performed on a
channel (clone channel) which was created with endpoint details
(ast_channel_alloc_with_endpoint()) and the original channel which is created
without endpoint details (ast_channel_alloc()) then both the channels must
exchange their endpoint details or else after masquerade when clone channel
is being destroyed the endpoint cleanup callbacks will be destroyed too and
after call completion unique_id of original channel will still be there in
ast_endpoint structure's channel_ids container.
This prevents use-after-scope issues when unwinding the stack,
which happens in reverse order. The varname variable needs to
remain alive for the destruction to be able to access it.
Issue was found using clang + address-sanitizer.
Alexei Gradinari [Tue, 18 Dec 2018 19:47:36 +0000 (14:47 -0500)]
res_pjsip: add option to enable ContactStatus event when contact is updated
The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out
the ContactStatus AMI event when a contact is updated.
Thist change broke things which rely on old behavior.
This patch adds a new PJSIP global configuration option
'send_contact_status_on_update_registration' to be able to preserve old
ContactStatus behavior.
By default new behavior, i.e. the ContactStatus event will not be sent when a
device refreshes its registration.
Joshua Colp [Mon, 7 Jan 2019 14:06:37 +0000 (14:06 +0000)]
res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled.
For video streams it was possible for the abs-send-time information
to be placed into RTP streams even if not negotiated. Depending on
the endpoint in use this could cause video to not flow.
We now only enable abs-send-time for negotiation if WebRTC is enabled.
RTP: reset DTMF last seqno/timestamp on RTP renegotiation
The remote side may start a new stream when renegotiating RTP.
Need to reset the DTMF last sequence number and the timestamp
of the last END packet on RTP renegotiation.
If the new time stamp is lower then the timestamp of the last DTMF END packet
the asterisk drops all DTMF frames as out of order.
This bug was caught using Cisco ip-phone SPA5XX and codec g722.
On SIP session update the SPA50X resets stream and a new timestamp is twice
smaller then the previous.
Bryan Boatright [Wed, 2 Jan 2019 17:44:41 +0000 (11:44 -0600)]
app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail
If a voicemail is marked "urgent" then the VM_MESSAGEFILE channel variable is
not updated correctly since urgent messages are in a different directory. The
fix is to update the channel variable when the path to the urgent message is
created.
Richard Mudgett [Wed, 19 Dec 2018 19:02:35 +0000 (13:02 -0600)]
stasic.c: Fix printf format type mismatches with arguments.
An int64_t is not likely the same size as a long.
* Changed the int64_t values in the statistics structs to longs so casting
is not necessary when generating the formatted CLI output. The offending
members did not need to be int64_t anyway as they were only set by an int
type variable which was already truncating bits.
* Reordered the statistics structs to reduce potential padding bytes.
George Joseph [Mon, 24 Dec 2018 17:42:36 +0000 (10:42 -0700)]
ast_coredumper: Refactor the pid determination process
In order to get a dump of the running process, we need to find the
pid of the main asterisk process. This can be tricky if there are
also instances of "asterisk -r" running or if an alternate location
for asterisk.conf was specified on the command line with the -C
option that also specified an alternation location for the pid file.
So now...
1. We find the asterisk executable with "which" or the --asterisk-bin
command line option.
2. If there's only 1 process with an executable path that matches,
we use that pid. If not...
3. We try "<asterisk-bin> -rx 'core show settings'" and parse the
output to find the pidfile, then read that for the pid. If that
didn't work...
4. We get a list of all the pids matching <asterisk-bin> and look
in /proc/<pid>/cmdline for a -C argument and retry the "core show
settings" using the same -C option. We can't parse the output
of "ps" to get the -C path because it may contain spaces. The
contents of /proc/<pid>/cmdline are delimited by NULLs. For BSDs
we may have to mount /proc first. :(
Richard Mudgett [Wed, 19 Dec 2018 18:39:08 +0000 (12:39 -0600)]
backtrace.c: Fix casting pointer to/from integral type.
The backtrace library bfd.h include file does not get the sizes of
pointers and ints right on some platforms. On my old test box the size
of bfd_vma is 8 while the size of a pointer is 4. gcc on the box
complains of the integer casting to/from pointers size mismatch.
* uintptr_t to the rescue by doing an appropriate two stage cast.
George Joseph [Tue, 18 Dec 2018 16:33:50 +0000 (09:33 -0700)]
app_voicemail: Don't delete mailbox state unless mailbox is deleted
The free_user function was automatically deleting the stasis mailbox
state but this only makes sense when the mailbox is actually
deleted, not just the structure freed. This was causing issues
where leave_voicemail would publish the mwi message to stasis and
delete the state before the message could be processed by
res_pjsip_mwi.
* Removed the delete of state from free_user().
* Created a new free_user_final() function that both frees the data
structure and deletes the state. This function is only called
during module load/unload where it's appropriate to delete the
state.
Sean Bright [Thu, 13 Dec 2018 21:56:50 +0000 (16:56 -0500)]
res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set
The profile-iop octet (the 2nd) of profile-level-id can be zero
according to RFC 6184 Section 8.1. So we ignore its value when deciding
to include profile-level-id in the outgoing SDP.
Joshua C. Colp [Fri, 30 Nov 2018 11:40:40 +0000 (07:40 -0400)]
stasis: Add statistics gathering in developer mode.
This change adds statistics gathering to Stasis topics,
subscriptions, and message types. These can be viewed using
CLI commands and provide insight into how Stasis is used
and how long certain operations take to execute.
These are only available when Asterisk is compiled in
developer mode and do not have any impact under normal
operation.
Sebastian Damm [Thu, 6 Dec 2018 17:23:50 +0000 (18:23 +0100)]
res/res_ari: Add additional hangup reasons
The ARI DELETE /channels command takes a "reason" parameter
Previously, there were only five reasons implemented
This patch adds more reasons to choose from for more
complex setups
Sean Bright [Fri, 7 Dec 2018 12:57:48 +0000 (07:57 -0500)]
utils: Wrap socket() and pipe() to reduce syscalls
Some platforms provide an implementation of socket() and pipe2() that allow the
caller to specify that the resulting file descriptors should be non-blocking.
Using these allows us to potentially elide 3 calls into 1 by avoiding extraneous
calls to fcntl() to set the O_NONBLOCK flag afterwards.
In passing, change ast_alertpipe_init() to use pipe2() directly instead of the
wrapper if it is available.
George Joseph [Thu, 29 Nov 2018 15:53:51 +0000 (08:53 -0700)]
stasis: Allow filtering by formatter
A subscriber can now indicate that it only wants messages
that have formatters of a specific type. For instance,
manager can indicate that it only wants messages that have a
"to_ami" formatter. You can combine this with the existing
filter for message type to get only messages with specific
formatters or messages of specific types.
George Joseph [Wed, 5 Dec 2018 15:37:45 +0000 (08:37 -0700)]
CI: Various updates to buildAsterisk.sh
* Added ---no-configure, --no-menuselect, --no-make and --no-alembic
options that prevent those actions from being performed. Useful
for testing and re-running portions of the build after fixing
earlier failures.
* Added "set -e" to abort the script on command failure.
Not sure why this wasn't there in the first place.
* Fixed a few echos that were redirecting to stderr when they shouldn't
have been.
* Catch more alembic failures by actually trying to generate the SQL.
WARNING[5812]: http.c:1939 httpd_helper_thread: Failed to set
TCP_NODELAY on HTTP connection: Bad file descriptor
ERROR[5812]: iostream.c:91 ast_iostream_nonblock: Failed to get
fcntl() flags for file descriptor: Bad file descriptor
ERROR[5812]: iostream.c:569 ast_iostream_close: close() failed: Bad
file descriptor
Disabled for now by making the test explicit only.
Pirmin Walthert [Wed, 28 Nov 2018 07:14:12 +0000 (08:14 +0100)]
pjproject_bundled: check whether UPDATE is supported on outgoing calls
In ASTERISK-27095 an issue had been fixed because of which chan_pjsip was not
trying to send UPDATE messages when connected_line_method was set to invite.
However this only solved the issue for incoming INVITES. For outgoing INVITES
(important when transferring calls) the options variable needs to be updated
at a different place.
That commit closed a long standing hole which allowed subscriptions
to mailboxes that weren't configured in voicemail.conf. This
caused an issue with FreePBX which depdended on that behavior.
The commit is being reverted until FreePBX can handle the new
behavior.
Corey Farrell [Mon, 26 Nov 2018 12:09:11 +0000 (07:09 -0500)]
jansson: Upgrade to 2.12.
This brings in jansson-2.12, removes all patches that were merged
upstream. README is created in third-party/jansson/patches to explain
how to add patches but also because the patches folder must exist for
the build process to succeed.
Alexei Gradinari [Fri, 23 Nov 2018 15:40:50 +0000 (10:40 -0500)]
RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit
The marker bit set on the voice packet indicates the start
of a new stream and a new time stamp.
Need to reset the DTMF last sequence number and the timestamp
of the last END packet.
If the new time stamp is lower then the timestamp of the last DTMF END packet
the asterisk drops all DTMF frames as out of order.
This bug was caught using Cisco ip-phone SPA50X and codec g722.
On SIP session update the SPA50X resets stream indicating it with market bit
and a new timestamp is twice smaller then the previous.
Corey Farrell [Wed, 14 Nov 2018 11:02:20 +0000 (06:02 -0500)]
astobj2: Create function to copy weak proxied objects from container.
Create ao2_container_dup_weakproxy_objs to perform a similar function to
ao2_container_dup. This function expects the source container to have
weakproxy objects, inserts the associated non-weak objects into the
destination container. Orphaned weakproxy objects are ignored.
Create test for this new function and for ao2_weakproxy_find.
Kevin Harwell [Fri, 16 Nov 2018 20:45:23 +0000 (14:45 -0600)]
func_strings: HASHKEY - negative array index can cause corruption
This patch makes it so only matching non-empty key names, and keys created by
the HASH function are eligible for inclusion in the comma separated string. It
also fixes a bug where it was possible to write to a negative index if the
result buffer was empty.
ASTERISK-28159
patches:
ASTERISK-28159.diff submitted by Michael Walton (license 6502)
George Joseph [Mon, 19 Nov 2018 17:59:07 +0000 (10:59 -0700)]
CI: Get job timeouts from environment
The job timeouts were hard coded in the jenkinsfiles which
means changes had to go through gerrit. Now they are taken
from the following environment variables (and their defaults) that
can be set in Jenkins configuration...
Joshua C. Colp [Sun, 18 Nov 2018 23:53:14 +0000 (19:53 -0400)]
stasis: Remove stringfields and lock from change message.
When a subscribe or unsubscribe occurs a message is published
containing this information. This change makes it so that the
message no longer uses stringfields or a lock, as both are not
really needed for the message.
This replaces the inline functions with macros. This removes the need
to directly use __ao2_ref, opts instead for standard ao2_bump and
ao2_cleanup macros.
George Joseph [Thu, 8 Nov 2018 15:53:44 +0000 (08:53 -0700)]
backtrace: Refactor ast_bt_get_symbols so it doesn't crash
We've been seeing crashes in libbfd when we attempt to generate
a stack trace from multiple threads. It turns out that libbfd
is NOT thread-safe. It can cache the bfd structure and give it to
multiple threads without protecting itself. To get around this,
we've added a global mutex around the bfd functions and also have
refactored the use of those functions to be more efficient and
to provide more information about inlined functions.
Also added a few more tests to test_pbx.c. One just calls
ast_assert() and the other calls ast_log_backtrace(). Neither are
run by default.
WARNING: This change necessitated changing the return value of
ast_bt_get_symbols() from an array of strings to a VECTOR of
strings. However, the use of this function outside Asterisk is not
likely.
Sungtae Kim [Sat, 17 Nov 2018 02:33:20 +0000 (03:33 +0100)]
res/res_ari: Fix null endpoint handle
The res_ari(POST /channels/create handler) deos not check the endpoint
parameter length. And it causes core
dump.
Fixed it to check the parameter length. Also fixed memory leak.
This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.
This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.
There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.
projects / thirdparty / asterisk.git / log
