Commit Graph

30250 Commits

Author SHA1 Message Date
Alexander Traud
78855e13e8 chan_oss: Compile in Solaris 11.
M_READ existed already and was conflicting in name.

Change-Id: I02108e07ae7d2dc314fe1e6c706c17731095a3e4
2018-06-21 04:17:51 -06:00
Kevin Harwell
e7ffbbd40e Merge "Fix some doxygen and curly placement." into 13 2018-06-20 17:15:35 -05:00
Jenkins2
8cc080a95c Merge "Dialplan functions: Fix some channel autoservice misuse." into 13 2018-06-20 16:36:58 -05:00
Jenkins2
8c3edf3a69 Merge "menuselect/menuselect_curses: Resolves sprintf usage error" into 13 2018-06-20 10:33:47 -05:00
Richard Mudgett
a6edbc9be1 Dialplan functions: Fix some channel autoservice misuse.
* Fix off nominal paths leaving the channel in autoservice.
* Remove unnecessary start/stop channel autoservice.
* Fix channel locking around a channel datastore search.

Change-Id: I7ff2e42388064fe3149034ecae57604040b8b540
2018-06-19 11:49:24 -05:00
Richard Mudgett
7b30158147 Fix some doxygen and curly placement.
Change-Id: I9a784a7c804120a8fa826c2a4cb9957e4b0b2fc8
2018-06-19 11:46:45 -05:00
Matthew Fredrickson
3570a62b32 menuselect/menuselect_curses: Resolves sprintf usage error
Acccording to the man page for sprintf, using the same buffer for
output as one used as an input yields undefined behavior.
This patch should work around this problem.

ASTERISK-27903
Reported-by: Alexander Traud

Change-Id: I2213dcb454aff26457e2e4cc9c6821276463ae3a
2018-06-19 08:43:31 -06:00
Richard Mudgett
f94a310ca0 channel: Fix some more unprotected channel flag setting.
Change-Id: I34c3b1201b1de539945bcfdcb264fff30332d48c
2018-06-18 10:54:03 -05:00
Sam Wierema
bb0ce22b2b app_mp3: remove 10 seconds of silence after mp3 playback
This patch changes the way asterisk polls output from mpg123, instead
of waiting for 10 seconds(when playing an http url) it now uses a
timeout of one second and iterates 10 times using this same timeout.

The main difference is that for every timeout asterisk receives it now
checks if mpg123 is still running before poll again.

ASTERISK-27752

Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620
2018-06-15 07:24:15 -06:00
Jenkins2
c72aef9657 Merge "tests/test_utils: Repair ./configure --with-ssl=PATH." into 13 2018-06-14 11:41:46 -05:00
Joshua Colp
4f5b8f77b9 Merge "res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP." into 13 2018-06-14 11:27:16 -05:00
Alexander Traud
b004b44a46 tests/test_utils: Repair ./configure --with-ssl=PATH.
ASTERISK-27914

Change-Id: Ibcab8f556ee77776f203cff8b06d776a673b7bc4
2018-06-14 00:38:19 -06:00
ktyerman
c6116a3234 chan_iax2: better handling for timeout and EINTR
The iax2 module is not handling timeout and EINTR case properly. Mainly when
there is an interupt to the kernel thread. In case of ast_io_wait recieves a
signal, or timeout it can be an error or return 0 which eventually escapes the
thread loop, so that it cant recieve any data. This then causes the modules
receive queue to build up on the kernel and stop any communications via iax in
asterisk.

The proposed patch is for the iax module, so that timeout and EINTR does not
exit the thread.

ASTERISK-27705
Reported-by: Kirsty Tyerman

Change-Id: Ib4c32562f69335869adc1783608e940c3535fbfb
2018-06-13 16:46:40 -06:00
Alexander Traud
e3de4bc46e res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP.
Previously, Asterisk used its script ./configure, to test whether OpenSSL was
built with no-srtp (or was simply too old). However, the header file
<openssl/opensslconf.h> is the preferred way to detect the local configuration
of OpenSSL.

As a positive side-effect the script ./configure does not interleave the
detection of the Open Settlement Protocol Toolkit (OSPTK) with the detection of
individual features of OpenSSL anymore.

Change-Id: I3c77c7b00b2ffa2e935632097fa057b9fdf480c0
2018-06-13 08:00:35 -06:00
Jenkins2
49c4c8af98 Merge "res_rtp_asterisk: Allow OpenSSL configured with no-deprecated." into 13 2018-06-12 10:06:44 -05:00
Joshua Colp
7609249a22 Merge "crypto.h: Repair ./configure --with-ssl=PATH." into 13 2018-06-12 09:40:01 -05:00
Joshua Colp
2b25fdb80c Merge "res_crypto: Allow OpenSSL configured with no-deprecated." into 13 2018-06-12 08:28:16 -05:00
Jenkins2
e46ab4743b Merge "res_srtp: Repair ./configure --with-ssl=PATH." into 13 2018-06-12 07:45:19 -05:00
Jenkins2
6fa33215ef Merge "func_odbc: NODATA if SQLNumResultCols returned 0 columns on readsql" into 13 2018-06-12 07:36:15 -05:00
Jenkins2
867a23e267 Merge "chan_pjsip: Register for "BEFORE_MEDIA" responses" into 13 2018-06-11 18:05:10 -05:00
Richard Mudgett
034a04af2a AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses.
When endpoint specific ACL rules block a SIP request they respond with a
403 forbidden.  However, if an endpoint is not identified then a 401
unauthorized response is sent.  This vulnerability just discloses which
requests hit a defined endpoint.  The ACL rules cannot be bypassed to gain
access to the disclosed endpoints.

* Made endpoint specific ACL rules now respond with a 401 unauthorized
which is the same as if an endpoint were not identified.  The fix is
accomplished by replacing the found endpoint with the artificial endpoint
which always fails authentication.

ASTERISK-27818

Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32
2018-06-11 10:26:37 -05:00
Alexander Traud
ef2386fcd7 res_rtp_asterisk: Allow OpenSSL configured with no-deprecated.
Furthermore, allow OpenSSL configured with no-dh. Additionally, this change
allows auto-negotiation of the elliptic curve/group for servers, not only with
OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519
(since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect.

ASTERISK-27910

Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537
2018-06-08 22:09:00 +02:00
Alexander Traud
2c3ad1e40d crypto.h: Repair ./configure --with-ssl=PATH.
ASTERISK-27908

Change-Id: Iac49d9f82faeb8a4611c6805906bd6d650b1b1d8
2018-06-08 05:02:51 -06:00
Alexander Traud
7763aa824e res_crypto: Allow OpenSSL configured with no-deprecated.
The header <openssl/rsa.h> had to be included explicitly.

ASTERISK-27906

Change-Id: I41743801eed998c039d73db7a0762d104a4f75b2
2018-06-08 11:06:44 +02:00
Alexander Traud
e62e069705 res_srtp: Repair ./configure --with-ssl=PATH.
ASTERISK-27905

Change-Id: Ibb7dc148a0048f4f9c3b12937ba4240dff0d15e2
2018-06-08 01:43:33 -06:00
Alexei Gradinari
d9d50396cf func_odbc: NODATA if SQLNumResultCols returned 0 columns on readsql
The functions acf_odbc_read/cli_odbc_read ignore a number of columns
returned by the SQLNumResultCols.
If the number of columns is zero it means no data.
In this case, a SQLFetch function has to be not called,
because it will cause an error.

ASTERISK-27888 #close

Change-Id: Ie0f7bdac6c405aa5bbd38932c7b831f90729ee19
2018-06-07 10:29:59 -04:00
George Joseph
98da1971e3 chan_pjsip: Register for "BEFORE_MEDIA" responses
chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant
it was not updating HANGUPCAUSE for 4XX responses.  If the remote end
sent a "180 Ringing", then a "486 Busy", the hangup cause was left at
"180 Normal Clearing".

* Removed chan_pjsip_incoming_response from the original session
  supplement (which was handling only "AFTER MEDIA") and added it to a
  new session supplement which accepts both "BEFORE_MEDIA" and
  "AFTER_MEDIA".

* Also cleaned up some cleanup code in load module.

ASTERISK-27902

Change-Id: If9b860541887aca8ac2c9f2ed51ceb0550fb007a
2018-06-07 07:56:00 -06:00
Alexander Traud
1205bcb69f ooh323c: GCC 8.1 warned about output truncated before terminating nul.
ASTERISK-27901

Change-Id: I5a8e894f4924ef52e3094f6870656a559d67f3d7
2018-06-07 06:20:36 -06:00
Joshua Colp
19f8f85689 Merge "pjsip_options: handle modification of qualify options in realtime" into 13 2018-06-06 11:21:38 -05:00
George Joseph
8b6c84787c Merge "pjsip_options: show/reload AOR qualify options using CLI" into 13 2018-06-06 10:10:40 -05:00
George Joseph
499867d006 Merge "app_confbridge: Add talking indicator for ConfBridgeList AMI response" into 13 2018-06-06 09:46:29 -05:00
Joshua Colp
b5a186d723 Merge "bridge_channel.c: Fix Deadlock when using Local channels and fax gateway" into 13 2018-06-06 05:46:28 -05:00
George Joseph
8dbd3d2b05 Merge "tcptls: Allow OpenSSL configured with no-dh." into 13 2018-06-05 14:22:35 -05:00
George Joseph
a938f49bc5 Merge "tcptls.h: Repair ./configure --with-ssl=PATH." into 13 2018-06-05 14:20:38 -05:00
George Joseph
93a85150e1 Merge "tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated." into 13 2018-06-05 13:01:08 -05:00
Joshua Colp
f17d09ae63 Merge "app_meetme: Fix manager event documentation for several events." into 13 2018-06-05 06:53:33 -05:00
George Joseph
db2413b446 app_sendtext: Allow content types other than text/plain
There was no real reason to limit the conteny type to text/plain other
than that's what it was limited to before.  Now any text/* content
type will be allowed for channel drivers that don't support enhanced
messaging and any type will be allowed for channel drivers that do
support enhanced messaging.

Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9
2018-06-04 13:19:52 -06:00
Pirmin Walthert
8b67e2bd14 bridge_channel.c: Fix Deadlock when using Local channels and fax gateway
ast_indicate is invoked with the bridge locked. As ast_indicate locks the
other end of the bridge as well this can lead to a deadlock in some situations.
(Especially when a different thread does the same in the reverse order).
This patch calls ast_indicate after unlocking the bridge which fixes the
deadlock. Calling ast_indicate with these parameters without locking the
bridge should be safe as this is done at different places without a
bridge lock.

ASTERISK-27094 #close
Reported-by: David Brillert

Change-Id: I5f86c1e2ce75b9929a36ab589b18c450e62ea35f
2018-06-01 14:24:59 -06:00
William McCall
9ff4779f03 app_confbridge: Add talking indicator for ConfBridgeList AMI response
When an AMI client connects, it cannot determine if a user was talking
prior to a transition in the user speaking state (which would generate
a ConfbridgeTalking event). This patch causes app_confbridge to track the
talking state and make this state available via ConfBridgeList.

ASTERISK-27877 #close

Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6
2018-06-01 05:34:06 +00:00
Joshua Colp
a385467a35 Merge "ast_coredumper: Fix output directory and variable precedence" into 13 2018-05-31 05:15:57 -05:00
Richard Mudgett
071232244a app_meetme: Fix manager event documentation for several events.
The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and
MeetmeTalkRequest AMI events were documented with sending out a Usernum
header when the User header was actually output.

* Change the online documentation to match reality.

ASTERISK-27873
ASTERISK-25261

Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39
2018-05-29 12:38:13 -05:00
Joshua Colp
369e611ac0 Merge "libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated." into 13 2018-05-29 12:07:39 -05:00
Alexander Traud
8c78337479 tcptls.h: Repair ./configure --with-ssl=PATH.
asterisk/tcptls.h was included (explicitly, implicitly, or transitively). Those
inclusions got replaced by forward declarations. As side effect, the inclusions
got completed.

ASTERISK-27878

Change-Id: I9d102728e30336d6522e5e4ae9e964013a0835f7
2018-05-28 17:32:15 +02:00
Alexei Gradinari
9ad3918acd pjsip_options: handle modification of qualify options in realtime
Currentrly pjsip_options code does not handle the situation when the
qualify options were changed in realtime database.
Only 'module reload res_pjsip' helps.

This patch add a check on contact add/update observers if the contact
qualify options are different than local aor qualify options.
If the qualify options were modified then synchronize
the pjsip_options AOR local state.

ASTERISK-27872

Change-Id: Id55210a18e62ed5d35a88e408d5fe84a3c513c62
2018-05-25 18:01:42 -04:00
Alexander Traud
6833c763c7 tcptls: Allow OpenSSL configured with no-dh.
Additionally, this change allows auto-negotiation of the elliptic curve/group
for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer.
This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a
side-effect.

ASTERISK-27876

Change-Id: I62c2aba4a630aefc231b71f646207e8c027d9497
2018-05-25 08:56:46 -06:00
Alexander Traud
204cc25a27 tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated.
ASTERISK-27874

Change-Id: Ica65113511c7a1c13f7988e7d9e7d9e7f3f620dd
2018-05-25 14:24:51 +02:00
Joshua Colp
7f318c3ab5 Merge "res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change" into 13 2018-05-24 14:55:59 -05:00
George Joseph
acfdfcd19e ast_coredumper: Fix output directory and variable precedence
The OUTPUTDIR variable in ast_debug_tools.conf.sample is now set
to "/tmp" instead of "/some/directory".

Variables set on the command line or that are already in the
environment now take predecence over variables set in the config files.

ASTERISK-27846
Reported by: Ted G

Change-Id: Ie8baec52d531886bf5849ec1d59bb59dc87ad387
2018-05-24 13:00:06 -06:00
Joshua Colp
bb33dafa8f Merge "tcptls: Repair ./configure --with-ssl=PATH." into 13 2018-05-24 06:07:18 -05:00
Joshua Colp
27a8189b0c Merge "channel.c: Fix off nominal channel allocation failure path." into 13 2018-05-24 05:15:50 -05:00