Many uses of stasis_unsubscribe in modules can be reached through unload.
These have been switched to stasis_unsubscribe_and_join.
Some subscription callbacks do nothing, for these I've created a noop
callback function in stasis.c. This is used by some modules that monitor
MWI topics in order to enable cache, since the callback does not become
invalid after dlclose it is safe to use stasis_unsubscribe on these, even
during module unload.
ASTERISK-25121 #close
Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
The res_hep_rtcp module was incorrectly including <pjsip.h>. This didn't need
to be included, as the module does not using PJPROJECT any fashion.
Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as
a dependency, this also meant that res_hep_rtcp will fail to compile on a
system without PJPROJECT.
This patch removes the include.
Thanks to Damien Wedhorn for pointing this out in #asterisk-dev.
ASTERISK-24236 #close
Reported by: Damien Wedhorn, Matt Jordan
Tested by: Damien Wedhorn
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Merged revisions 421064 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 421065 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These modules were originally specified as being disabled, as they were
introduced midstream in Asterisk 12. That makes it nicer for folks who are
upgrading to a new release in the middle of Asterisk 12. That's not the case
for Asterisk 13: it's a brand new release. There's no reason to have the
modules disabled by default in that case.
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Merged revisions 420742 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new module to Asterisk, res_hep_rtcp. The module subscribes
to the RTCP topics in Stasis and receives RTCP information back from the
message bus. It encodes into HEPv3 packets and sends the information to the
res_hep module for transmission.
Using this, someone with a Homer server can get live call quality monitoring
for all RTP-based channels in their Asterisk 12+ systems.
In addition, there were a few bugs in the RTP engine, res_rtp_asterisk, and
chan_pjsip that were uncovered by the tests written for the Asterisk Test
Suite. This patch fixes the following:
1) chan_pjsip failed to set its channel unique ids on its RTP instance on
outbound calls. It now does this in the appropriate location, in the
serialized call callback.
2) The rtp_engine was overflowing some values when packed into JSON.
Specifically, some longs and unsigned ints can't be be packed into integer
values, for obvious reasons. Since libjansson only supports integers,
floats, strings, booleans, and objects, we print these values into strings.
3) res_rtp_asterisk had a few problems:
(a) it would emit a source IP address of 0.0.0.0 if bound to that IP
address. We now use ast_find_ourip to get a better IP address, and
properly marshal the result into an ast_strdupa'd string.
(b) Reports can be generated with no report bodies. In particular, this
occurs when a sender is transmitting information to a receiver (who
will send no RTP back to the sender). As such, the sender has no report
body for what it received. We now properly handle this case, and the
sender will emit SR reports with no body. Likewise, if we receive an
RTCP packet with no report body, we will still generate the appropriate
events.
ASTERISK-24119 #close
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Merged revisions 419823 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Corey Farrell
Rodrigo Ramírez Norambuena
Matt Jordan