OpenSSL 1.1.0 includes some major changes in the interface. See
https://wiki.openssl.org/index.php/1.1_API_Changes .
Status: Right now there are still a few deprecation notes with OpenSSL
1.1.0. But it's a start.
Changes:
* CRYPTO_LOCK is no longer available. Replace it with its value for now.
I don't completely understand what it is used for there.
* Remove several functions from libasteriskssl that seem to no longer be
needed.
* Structures have become opaque and are accesses with accessors.
* ERR_remove_thread_state() no longer needed.
* SSLv2 code now could no longer be used in 1.1.
ASTERISK-26109 #close
Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b
The sending codec is switched to the receiving codec and then
is switched back to the best native codec on EVERY receiving RTP packets.
This is because after call of ast_channel_set_rawwriteformat there is call
of ast_set_write_format which calls set_format which sets rawwriteformat
to the best native format.
This patch adds a new function ast_set_write_format_path which set
specific write path on channel and uses this function to switch
the sending codec.
ASTERISK-26603 #close
Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d
The asterisk.h header file needs to be included first or else
some things go awry, such as:
implicit declaration of function 'vasprintf'
Change-Id: I981dc2a77a1ba791888e4f1726644d4656c0407c
If a TCP/TLS connection was pending (not accepted and not timed out) during
unload of chan_sip, Asterisk would segfault when trying to send a signal to
a thread whose thread ID hadn't been recorded yet. This commit fixes that by
recording the thread ID before calling the blocking connect() syscall.
This was a regression introduced by 776a14386a.
The above wasn't enough to fix the segfault, which was now delayed to the
point where connect() timed out. Therefore, it was necessary to also remove
the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be
used to interruput the connect() syscall.
This was a regression introduced by 5d313f51b9.
ASTERISK-26586 #close
Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b
When retrieving RTCP stats for PJSIP channels, RTT values are unreliable.
RTT calculation is correct, but the data representation isn't. RTT is
represented by a 32-bit fixed-point number with the integer part in the
first 16 bits and the fractional part in the last 16 bits. In order to
get the RTT value, the fractional part is miscalculated, there is an
unnecessary 16 bit shift that causes overflow. Besides this there is
another mistake, when transforming the integer value to the fixed point
fractional part via bitwise operation, that loses precision.
* RTT fractional part is no longer shifted, avoiding overflow.
* RTT fractional part is transformed to its fixed-point value more
precisely.
* Fixed timeval2ntp() and ntp2timeval() second fraction conversions.
* Fixed NTP timestamp report logging. The usec was inexplicably
multiplied by 4096.
ASTERISK-26566 #close
Reported by Hector Royo Concepcion
Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f
libasteriskpj was hard coded to use -lrt but librt is linux specific
so we now use the LIB_RT variable which gets set by configure.
Change-Id: I41148884517e3031f7675a413d524c86e8614694
Fix support of OS's like openBSD that use an older nameser.h,
this change reverts the defines to the older style which on other
systems is found in nameser_compat.h
Tested on openBSD 6.0, Debian 8
ASTERISK-26608 #close
Change-Id: Iffb36caab8c5aa9dece0ce2d009041f7b56cc86a
In order to not have version number overlap between different versions
of Asterisk, each new major version of Asterisk will mean we also bump
the ARI major version number.
This particular change does NOT introduce any known breaking changes to
ARI.
For discussion relating to this topice, see:
http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html
Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665
The download process now has a timeout which will cause wget to retry
if it stops retrieving data for 5 seconds and fetch and curl to timeout
if the whole retrieval take smore than 30 seconds.
If the tarball retrieval works, the MD5SUM file is retrieved from
the downloads site and the md5 checksum is verified.
If either the tarball retrieval or MD5SUM retrieval fails, or the
checksums don't match, the entire process is retried once. If it
fails again, any incomplete tarball is deleted.
.DELETE_ON_ERROR: was also added to the Makefile. Not only does
this delete the tarball on failure, it till also delete corrupted
library files from the pjproject source directory should they
fail to build correctly.
Tested all the way back to FreeBSD 9, CentOS 6, Debian 6 and
Ubuntu 14.
Change-Id: Iea7d33b96a31622ab1b6e54baebaf271959514e1
POSIX defines poll.h. sys/poll.h should not be used as it is c-library
internal header which may or may not exist. Notably in musl including
sys/poll.h generates warning of being incorrect.
Change-Id: Ib318c1c7142a737bcf3caa4d8d72560bebe39252
OpenBSD's 'find' doesn't take the -delete argument so you have to pipe
through 'xargs rm -rf'.
'echo -e' doesn't like \t starting a line. It just prints 't' which
causes the libasteriskpj.exports file to be garbage. They were just
cosmetic so they were removed.
librt doesn't exist so the link of libasteriskpj.so fails. It's not
actually needed for linux anyway so -lrt was removed from the link.
res_rtp_asterisk was failing to load because of an undefined
DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if
so DTLSv1_method is used instead.
ASTERISK-26608
Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c
res_format_attr_opus assumed that the string being passed into it was
empty. It tried to determine if the only thing it had written was
a=fmtp:<num>
And if it had, it would reset the string. Its calculation was off when
working with chan_sip, though. chan_sip passes the entire built SDP
rather than an empty string. This resulted in always putting an empty
fmtp line in the SDP.
ASTERISK-26520 #close
Reported by scgm11
Change-Id: Ib2e8712d26a47067e5f36d5973577added01dbb5
When Opus is negotiated but not loaded, the log is spammed with messages
because the system does not know how to calculate the number of samples in
a frame.
* Suppress the warning by supplying a function that assumes 20ms of
samples in the frame. For pass through support it doesn't really seem to
matter what number of samples is returned anyway.
ASTERISK-26605 #close
Change-Id: Icf2273692f040dc2c45b01e72a790d11092f9e0f
Responding to authentication challenges leaks PJSIP memory pools.
The leak was introduced with a pjproject 2.5.5 API change.
https://trac.pjsip.org/repos/ticket/1929 changed the API usage of
pjsip_auth_clt_init() to require the new API pjsip_auth_clt_deinit() to
clean up cached authentication allocations that get allocated with
pjsip_auth_clt_reinit_req().
ASTERISK-26516 #close
Change-Id: I4473141b8c3961d0dc91c382beb3876b3efb45c8
One of the code paths in __ast_file_read_dirs will only get executed if
the OS doesn't support dirent->d_type OR if the filesystem the
particular file is on doesn't support it. So, while standard Linux
systems support the field, some filesystems like XFS do not. In this
case, we need to call stat() to determine whether the directory entry
is a file or directory so we append the filename to the supplied
directory path and call stat. We forgot to truncate path back to just
the directory afterwards though so we were passing a complete file name
to the callback in the dir_name parameter instead of just the directory
name.
The logic has been re-written to only create a full_path if we need to
call stat() or if we need to descend into another directory.
Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba
