This reverts commit 94a4eea7f6.
This commit caused issues with polling when combined with
the revert commit "Revert "app_voicemail: Remove need to subscribe
to stasis"
ASTERISK-28222
Reported by: abelbeck
Change-Id: I0aaab5218a282706dfc6fcbbaeead228d599b99b
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. :(
ASTERISK-28221
Reported by: Andrew Nagy
Change-Id: I8aa1f3f912f949df2b5348908803c636bde1d57c
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.
ASTERISK-28162 #close
Change-Id: Ic72b4497e74d801b27a635559c1cf29c16c95254
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.
Change-Id: Icb2621583f50c8728de08a3c824d95fe53cc45d0
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.
ASTERISK-28215
Change-Id: I305e8b3c930e9ac41d901e5dc8a58fd7904d98dd
All of the fields that were removed were no longer referenced except for
'lastrxts' and 'rxseqno' which were only ever written to.
Change-Id: I5a5d31eb33e97663843698f58d0d97f22a76627c
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.
ASTERISK-27959 #close
Reported by: David Kuehling
Change-Id: Id28cd916a3d7748058fe9609b585d07d9e243f73
This is how features behaved up through Asterisk 11, but was changed
when the new bridging framework was implemented in Asterisk 12.
Reported by rrittgarn in #asterisk.
Change-Id: I72cf86223947a8118c75f46e2c603dbc11e3125b
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.
ASTERISK-28117
Change-Id: I94411b53767f89ee01714daaecf0c2f1666e863f
Currently the file sound_only_person is not played when a marked
user (with announce_only_user=yes) joins an empty conference.
This patch fixes it.
ASTERISK-28201 #close
Change-Id: I85b67687e6b220939c3af8091d83a70a7b174cf4
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.
ASTERISK-28186
Change-Id: Ifdb7a222a73b6b56c6bb9e4ee93dc8a394a5494c
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.
Change-Id: I3ebe654fb549587537161506c6c950f4ab298bb0
* 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.
Change-Id: I9f395fa4e9254be7299e7c1014f1a13db78faffb
This adds documentation to handle_cli_malloc_trim() indicating how it
can be useful when debugging OOM conditions.
Change-Id: I1936185e78035bf123cd5e097b793a55eeebdc78
We've had multiple opportunities where Richard Mudgett's
malloc_trim patch has been useful. Let's get it
pushed up to gerrit and merged.
Since malloc_trim is only available in libc, an entry is
added to configure.ac to create a definition for
HAVE_MALLOC_TRIM.
Change-Id: Ia38308c550149d9d6eae4ca414a649957de9700c
This test was occasionally failing, with:
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.
Change-Id: I778f6cbb6104c6b4e89737a2eaf1a9540888d351
This reverts commit 1843b0e2b5.
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.
ASTERISK-28151
Reported by: Ronald Raikes
Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15
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.
ASTERISK-28182 #close
Reported-by: nappsoft
Change-Id: I76cc06da4ca76ddd6dce814a8b97cc66b98aaf29
These are only a few of the leaks. The large number of macros
and return paths in this file would make a weeks worth of work
to plug them all.
Change-Id: Ie2369fa944023d44767871c5c30974cb077ffb56
