Commit Graph

28976 Commits

Author SHA1 Message Date
George Joseph
0bde568669 pjproject_bundled: Use the asterisk github mirror for download
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
2017-06-13 09:47:43 -06:00
Joshua Colp
bc51d4324a Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing." into 13 2017-06-13 09:18:18 -05:00
Alexei Gradinari
08be5e01e8 app_voicemail: IMAP logout on MWI unsubscribe
Closing IMAP connection on MWI unsubscribe.

ASTERISK-24052 #close

Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd
2017-06-12 18:55:15 -04:00
Joshua Colp
d715beae16 Merge "codecs.conf.sample: Fix max_bandwidth speling error" into 13 2017-06-12 16:02:00 -05:00
Joshua Colp
052cd049bc Merge "BuildSystem: Fix build on FreeBSD due to missing crypt.h" into 13 2017-06-12 15:57:32 -05:00
Jenkins2
1f7aedf6f7 Merge "eventfd: Disable during cross compilation" into 13 2017-06-12 15:31:51 -05:00
Alexei Gradinari
59c9bbe696 res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled
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
2017-06-12 10:57:24 -04:00
David M. Lee
68de35a6a0 CFLAGS for BIND8 support
Some systems (like macOS) require BIND_8_COMPAT to be defined so that
the nameser libraries are, well, BIND8 compatible.

Change-Id: If79fc27a64f90de1835b5aa3aadfa9be22bd16b0
2017-06-12 08:46:00 -05:00
Sean Bright
da3312457e codecs.conf.sample: Fix max_bandwidth speling error
Reported by Sylvain Boily via asterisk-dev mailing list.

Change-Id: Idc7623f335aea3e144dd369ba383b9a757480a9d
2017-06-11 13:06:17 -04:00
Guido Falsi
6a64f65fe6 BuildSystem: Add patches to allow building with recent LibreSSL
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
2017-06-08 17:54:46 +02:00
Jenkins2
85dff8e26e Merge "CHANGES: correct version for a new option 'refer_blind_progress'" into 13 2017-06-08 10:48:13 -05:00
Guido Falsi
44cee2f4a1 BuildSystem: Fix build on FreeBSD due to missing crypt.h
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
2017-06-08 17:36:00 +02:00
Joshua Colp
1f10c6b3b0 chan_pjsip: Update device state when in early media.
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
2017-06-07 20:19:05 +00:00
Sean Bright
590ffcaf0b eventfd: Disable during cross compilation
Reported by Lonnie Abelbeck <lonnie@abelbeck.com> via private e-mail.

Change-Id: Icc80f12b8d8d591e14a8e0ed9f1c02cbd193a89b
2017-06-07 14:36:17 -05:00
Alexei Gradinari
5520b6c201 CHANGES: correct version for a new option 'refer_blind_progress'
Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97
2017-06-07 12:21:10 -04:00
Joshua Colp
996a4791ff pjsip: Extend 'asymmetric_rtp_codec' option to include us changing.
PJSIP support in Asterisk differs from chan_sip in that it
allows media to be sent as-is without transcoding provided
the codecs were negotiated in the SDP. This is allowed
according to the RFC. Support for this differs quite a lot
though and some endpoints do not handle it well.

This change extends the 'asymmetric_rtp_codec' option to
also cover this case. When set to no (the default) the code
behaves as chan_sip does - the best codec is selected and
we will only ever send that, unless we change what we are
sending if the remote side changes. When set to yes we
will send media as-is without transcoding if the codec
has been negotiated in the SDP.

ASTERISK-26996

Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51
2017-06-07 13:12:55 +00:00
Jenkins2
812f5b51cb Merge "res_pjsip: Add support for returning only reachable contacts and use it." into 13 2017-06-07 08:11:23 -05:00
Jenkins2
f0400ed858 Merge "channel: ast_write frame wrongly freed after call to audiohooks" into 13 2017-06-07 07:58:41 -05:00
Sean Bright
c093bf8072 res_rtp_multicast: Use consistent timestamps when possible
When a frame destined for a MulticastRTP channel does not have timing
information (such as when an 'originate' is done), we generate the RTP
timestamps ourselves without regard to the number of samples we are
about to send.

Instead, use the same method as res_rtp_asterisk and 'predict' a
timestamp given the number of samples. If the difference between the
timestamp that we generate and the one we predict is within a specific
threshold, use the predicted timestamp so that we end up with timestamps
that are consistent with the number of samples we are actually sending.

Change-Id: I2bf0db3541b1573043330421cbb114ff0f22ec1f
2017-06-06 11:54:07 -04:00
Joshua Colp
746c2c5745 res_pjsip: Add support for returning only reachable contacts and use it.
This introduces the ability for PJSIP code to specify filtering flags
when retrieving PJSIP contacts. The first flag for use causes the
query code to only retrieve contacts that are not unreachable. This
change has been leveraged by both the Dial() process and the
PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt
calls to contacts which are not unreachable.

ASTERISK-26281

Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c
2017-06-06 14:45:49 +00:00
Jenkins2
95b2b542f6 Merge "format: Reintroduce smoother flags" into 13 2017-06-06 08:49:35 -05:00
Joshua Colp
3d7e9375fb Merge "res_srtp: Add support for libsrtp2" into 13 2017-06-06 05:00:57 -05:00
Kevin Harwell
adfb28882b channel: ast_write frame wrongly freed after call to audiohooks
ASTERISK-26419 introduced a bug when calling ast_audiohook_write_list in
ast_write. It would free the frame given to ast_write if the frame returned
by ast_audiohook_write_list was different than the given one. The frame
give to ast_write should never be freed within that function. It is the
caller's resposibility to free the frame after writing (or when it its done
with it). By freeing it within ast_write this of course led to some memory
corruption problems.

This patch makes it so the frame given to ast_write is no longer freed within
the function. The frame returned by ast_audiohook_write_list is now subsequently
used in ast_write and is freed later. It is freed either after translate if the
frame returned by translate is different, or near the end of ast_write prior
to function exit.

ASTERISK-26973 #close

Change-Id: I463d4ac3b736ced95de986ee74a489c7c7ab103b
2017-06-05 10:45:25 -05:00
Jenkins2
cee5b78be8 Merge "pbx_builtin: Properly handle hangup during Background" into 13 2017-06-05 07:35:23 -05:00
Jenkins2
174a9b5d99 Merge "stasis_recording: Correct ast_asprintf error checking" into 13 2017-06-01 10:47:53 -05:00
Jenkins2
1e086cf018 Merge "format_mp3: Re-work menuselect/build issues" into 13 2017-06-01 10:16:20 -05:00
Jenkins2
3e8eea0325 Merge "res_pjsip: New endpoint option "refer_blind_progress"" into 13 2017-06-01 09:48:48 -05:00
Jenkins2
a76b473c49 Merge "app_confbridge: Race between removing and playing name recording while leaving" into 13 2017-06-01 09:17:44 -05:00
Joshua Colp
b276810aa1 Merge "sip.conf.sample: Clarify where DTLS settings are permitted" into 13 2017-06-01 08:32:57 -05:00
Jenkins2
3f336ca053 Merge "test_json: Fix test names with reserved words" into 13 2017-05-31 13:59:39 -05:00
Sean Bright
283cc59af7 pbx_builtin: Properly handle hangup during Background
Before this patch, when a user hung up during a Background, we would
stuff 0xff into a char and attempt a dialplan lookup of it. This caused
problems for some realtime engines which interpreted the value as the
beginning of an invalid UTF-8 sequence.

ASTERISK-19291 #close
Reported by: Andrew Nowrot

Change-Id: I8ca6da93252d61c76ebdb46a4aa65e73ca985358
2017-05-31 13:22:12 -04:00
Joshua Colp
dc05183f4b channel / app_meetme: Fix parentheses.
ASTERISK-27025

Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed
2017-05-31 13:59:04 +00:00
Sean Bright
cf6cf59646 stasis_recording: Correct ast_asprintf error checking
ASTERISK-27021 #close
Reported by: Tim Morgan

Change-Id: I0ac061f040093e806c3b1f4e2340864f3ce4dd75
2017-05-30 17:07:56 -04:00
Sean Bright
70e5887906 format: Reintroduce smoother flags
In review 4843 (ASTERISK-24858), we added a hack that forced a smoother
creation when sending signed linear so that the byte order was adjusted
during transmission. This was needed because smoother flags were lost
during the new format work that was done in Asterisk 13.

Rather than rolling that same hack into res_rtp_multicast, re-introduce
smoother flags so that formats can dictate their own options.

Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16
2017-05-30 15:08:05 -05:00
Sean Bright
97b003f5e2 format_mp3: Re-work menuselect/build issues
Rather than removing format_mp3 from ALL_C_MODS (which caused format_mp3
to not show up in menuselect), use .PHONY targets when the necessary
source files are not present.

ASTERISK-23951
Reported by: Tzafrir Cohen

Change-Id: I0a7512c51acc9e86043671795020b0de725bd9e8
2017-05-30 10:51:00 -04:00
George Joseph
c10341646d test_json: Fix test names with reserved words
Some of the test names were actually reserved words (true, false,
int, null, string, bool).  When the jenkins test results analyzer
does its thing it tries to create a map using the test names as
keys and fails because they're reserved words.

Added "type_" to those test names.

Change-Id: I90d809f46969c78a1c605b736ff0635196a2cf1b
2017-05-30 08:43:49 -06:00
Joshua Colp
1e65857e6f Merge "format_mp3: Don't try to build format_mp3 if we don't have sources" into 13 2017-05-30 05:53:33 -05:00
Joshua Colp
b07b216235 manager: Clear the flag on the other channel.
During the channel flag audit an incorrect change was
done. The flag should be cleared on the second channel.

ASTERISK-26469

Change-Id: I770c5a389550a2fb5a6ade942fccbb2e1d9199c8
2017-05-26 16:41:59 +00:00
Sean Bright
5e9cd1f20d res_srtp: Add support for libsrtp2
ASTERISK-25294 #close
Reported by: Tzafrir Cohen

ASTERISK-26976 #close
Reported by: Alex

Change-Id: I789b1c3d1ed31365bbd9339fa58ef36f48833c40
2017-05-26 12:06:34 -04:00
Jenkins2
d4ccd3a6c0 Merge "asterisk: Audit locking of channel when manipulating flags." into 13 2017-05-26 09:12:11 -05:00
Jenkins2
5715360ba5 Merge "res_agi: Fix malformed AGI usage response" into 13 2017-05-26 08:00:49 -05:00
George Joseph
a8f8c5d857 Merge "res_agi: Allow configuration of audio format of EAGI pipe" into 13 2017-05-25 19:01:19 -05:00
George Joseph
a2d15b93f1 Merge "unittests: Add a unit test that causes a SEGV and..." into 13 2017-05-25 15:06:15 -05:00
Jenkins2
558199e5dd Merge "res_agi: Prevent crash when SET VARIABLE called without arguments" into 13 2017-05-25 14:44:11 -05:00
Sean Bright
72213c98e3 format_mp3: Don't try to build format_mp3 if we don't have sources
ASTERISK-23951 #close
Reported by: Tzafrir Cohen

Change-Id: Iebf181d44bb735787fde4b5be863c4d7e2478a30
2017-05-25 12:13:48 -04:00
Jenkins2
a3684b74e6 Merge "res_agi: Clarify 'RECORD FILE' documentation" into 13 2017-05-24 17:58:57 -05:00
George Joseph
65898c3af8 unittests: Add a unit test that causes a SEGV and...
...that can only be run by explicitly calling it with
'test execute category /DO_NOT_RUN/ name RAISE_SEGV'

This allows us to more easily test CI and debugging tools that
should do certain things when asterisk coredumps.

To allow this a new member was added to the ast_test_info
structure named 'explicit_only'.  If set by a test, the test
will be skipped during a 'test execute all' or
'test execute category ...'.

Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed
2017-05-24 14:56:14 -06:00
Joshua Colp
1bddf1efc3 Merge "chan_sip: Better ICE handling for RTCP-MUX" into 13 2017-05-24 11:41:30 -05:00
Jenkins2
a69905af69 Merge "res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm" into 13 2017-05-24 11:12:11 -05:00
Jenkins2
4dfcccdb70 Merge "res_format_attr_h26x: Trim blanks in fmtp attributes" into 13 2017-05-24 09:39:40 -05:00