build_tools/make_version: Get MAINLINE_BRANCH from .gitreview.
Use .gitreview defaultbranch setting to determine the mainline branch.
This allows the script to be used against other directories which might
not be on the same defaultbranch. This can be used by CI scripts to
report the testsuite version being used:
./build_tools/make_version ${TESTSUITE_DIR}
sched: Make ABI compatible between dev mode and non-dev mode.
In the past there was an assertion in the ast_sched_del function
and in order to ensure it was useful the calling function name,
line number, and filename had to be passed in. This cause the ABI
to be different between dev mode and non-dev mode.
This assertion is no longer present so the special logic can be
removed to make it the same between them both.
res_pjsip: Update default keepalive interval to 90 seconds.
A change recently went in which disabled the built-in PJSIP
keepalive. This defaulted to 90 seconds and kept TCP/TLS
connections alive. Disabling this functionality has resulted
in a behavior change of not doing keepalives by default resulting
in TCP/TLS connections dropping for some people.
This change makes our default keepalive interval 90 seconds
to match the previous behavior and preserve it.
George Joseph [Fri, 20 Jul 2018 11:20:25 +0000 (05:20 -0600)]
xmldoc.c: Fix dump of xml document
The "xmldoc dump" cli command was simply concatenating xml documents
into the output file. The resulting file had multiple "xml"
processing instructions and multiple root elements which is illegal.
Normally this isn't an issue because Asterisk has only 1 main xml
documentation file but codec_opus has its own file so if it's
downloaded and you do "xmldoc dump", the result is invalid.
* Added 2 new functions to xml.c:
ast_xml_copy_node_list creates a copy of a list of children.
ast_xml_add_child_list adds a list to an existing list.
* Modified handle_dump_docs to create a new output document and
add to it the children from each input file. It then dumps the
new document to the output file.
Previously, Asterisk did not tell its bundled PJProject about this configure
parameter. Therefore, PJProject used the platform provided OpenSSL always.
Ben Ford [Thu, 10 May 2018 18:11:06 +0000 (13:11 -0500)]
res_rtp_asterisk: Add support for sending NACK requests.
Support has been added for receiving a NACK request and handling it.
Now, Asterisk can detect when a NACK request should be sent and knows
how to construct one based on the packets we've received from the remote
end. A buffer has been added that will store out of order packets until
we receive the packet we are expecting. Then, these packets are handled
like normal and frames are queued to the core like normal. Asterisk
knows which packets to request in the NACK request using a vector
which stores the sequence numbers of the packets we are currently missing.
If a missing packet is received, cycle through the buffer until we reach
another packet we have not received yet. If the buffer reaches a certain
size, send a NACK request. If the buffer reaches its max size, queue all
frames to the core and wipe the buffer and vector.
According to RFC3711, the NACK request must be sent out in a compound
packet. All compound packets must start with a sender or receiver
report, so some work was done to refactor the current sender / receiver
code to allow it to be used without having to also include sdes
information and automatically send the report.
Also added additional functionality to ast_data_buffer, along with some
testing.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
res_sorcery_config: Allow configuration section to be used based on name.
A problem I've seen countless times is a global or system section
for PJSIP not getting applied. This is inevitably the result of
the "type=" line missing. This change alleviates that problem.
The ability to specify an explicit section name has been
added to res_sorcery_config. If the configured section
name matches this and there are no unknown things configured
the section is taken as being for the given type.
Both the PJSIP "global" and "system" types now support this
so you can just name your section "global" or "system" and it
will be matched and used, even without a "type=" line.
stasis: Improve message type "Use of before/init after destruction"
Fixes issue where error msg
"Use of before/init after destruction"
was being printed on disabled messages
in dev mode. With this
fix if message is disabled
a warning will print.
Nick French [Tue, 17 Jul 2018 14:09:04 +0000 (09:09 -0500)]
res_pjsip: Remove spurious error logging when printing silent headers
Asterisk patched the pjproject source to avoid crashing when pjproject
sip_msg headers are encountered with NULL vptr's, but the patch also
output error messages for some valid headers which simply did not need
to be added to the message itself, such as hidden route headers.
pjproject has since applied a similar patch to their baseline to avoid
crashes, but their version also avoids the spurious error logging.
Richard Mudgett [Fri, 13 Jul 2018 23:26:46 +0000 (18:26 -0500)]
Build: Fix modules getting their optimization setting overridden.
Asterisk modules that use PJPROJECT services have their compiler
optimization and possibly their symbolic debug options overridden by the
PJPROJECT configure script selected settings.
* We need to filter-out any -O and -g options in PJ_CFLAGS before echoing
out the result so the PJPROJECT_INCLUDE variable does not override the
Asterisk module settings when using bundled PJPROJECT.
NOTE: This patch only has an effect when using bundled PJPROJECT.
Turn off the periodic sending of CRLNCRLN. Default is on (90 seconds),
which conflicts with the global section's keep_alive_interval option in
pjsip.conf.
patches:
pjsip_keep_not_alive.patch submitted by Alexander Traud (License 6520)
George Joseph [Wed, 11 Jul 2018 11:14:49 +0000 (05:14 -0600)]
CI: Initial commit for moving CI into source repo
Create tests/CI directory and add files used by Jenkins to
build and test Asterisk.
With this commit, Jenkins will run the Asterisk Unit Tests using
the Jenkinsfile at tests/CI/unittests.jenkinsfile. Bash scripts
to do the actual building and testing are also in the same directory.
Output is placed in tests/CI/output so that directory has been
added to .gitignore.
George Joseph [Tue, 10 Jul 2018 18:28:09 +0000 (12:28 -0600)]
app_confbridge: Use the SDP 'label' attribute to correlate users
Previously, the msid "label" attribute was used to correlate
participant info but because streams could be reused, the msid
wasn't being updated correctly when someone left the bridge and
another joined.
Now, instead of looking for the msid attribute on a channel's streams,
app_confbridge sets an "SDP:LABEL" attribute on the stream which
res_pjsip_sdp_rtp looks for. If it finds it, it adds a "label"
attribute to the current sdp.
Kevin Harwell [Fri, 6 Jul 2018 20:05:47 +0000 (15:05 -0500)]
res_pjsip_session: sdp group:BUNDLE attribute being truncated
When setting/appending the media id's to the bundle group attribute a '-1' was
being passed to the 'ast_str_set/append' function for the 'max_len' parameter.
This essentially capped the length of the string to what it was originally
allocated with. In this case 64 bytes.
This patch makes it so a '0' is passed as in for the 'max_len', which means
"no maximum length".
res_pjsip_pubsub: segfault in function publish_expire
The function pubsub_on_rx_publish_request incorrectly uses
of AST_SCHED_REPLACE_UNREF.
The AST_SCHED_REPLACE_UNREF should unref old '_data'.
Because of this, there may be a double unref
of variable 'publication' when ast_sched_del is unsuccessful
that leads to use after free of the 'publication' in publish_expire.
George Joseph [Fri, 6 Jul 2018 14:04:56 +0000 (08:04 -0600)]
test.c: Make output jUnit compatible
Separate "name" into "classname" and "name".
Use '.' for classname separator instead of '/'.
Prefix reserved words with '_'.
Wrap output with a top-level "testsuites" element.
George Joseph [Fri, 6 Jul 2018 12:57:37 +0000 (06:57 -0600)]
res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint
A new option 'suppress_q850_reason_headers' has been added to the
endpoint object. Some devices can't accept multiple Reason headers and
get confused when both 'SIP' and 'Q.850' Reason headers are received.
This option allows the 'Q.850' Reason header to be suppressed.
The default value is 'no'.
ASTERISK-27949 Reported-by: Ross Beer
Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1
res_pjsip_t38: Decline T.38 stream on failure case.
When negotiating an incoming T.38 stream the code incorrectly
returned failure instead of a decline for the stream when a
problem occurred or the configuration didn't allow it. This
resulted in SDP offers being rejected with a 488 response
in all cases, even when another valid stream was present.
This change makes it so the stream is now declined. If no
streams are accepted a 488 response is sent while if at least
one stream is accepted all the declined streams are, well,
declined.
Richard Mudgett [Mon, 2 Jul 2018 23:43:10 +0000 (18:43 -0500)]
res_pjsip_t38.c: Be smarter about how we respond when T.38 is disabled.
We were blindly responding with AST_T38_REFUSED when ANY T.38 control
frame came accross the bridge. This causes T.38 Gateway to get confused
and the T.38 session to get in a strange state.
* Made the T.38 framehook only respond to request frames and ignore
response frames.
Richard Mudgett [Tue, 3 Jul 2018 17:10:36 +0000 (12:10 -0500)]
res_pjsip/pjsip_transport_management.c: Fix deadlock with transport keep alive.
Using the keep_alive_interval option can result in a deadlock between the
pjproject transport manager group lock and the monitored transports ao2
container lock. The pjproject transport manager group lock has to be
superior in the locking order to the monitored transports ao2 container
lock because of pjproject callbacks called when already holding the group
lock. The lock inversion happens when Asterisk attempts to send a keep
alive packet over the reliable transports.
* Made keepalive_transport_thread() iterate over the monitored transports
container rather than use the ao2_callback() method. This avoids holding
the container lock when sending the keep alive packet.
Richard Mudgett [Fri, 29 Jun 2018 23:28:26 +0000 (18:28 -0500)]
res_pjsip_t38.c: Fix crash by ignoring 1xx messages.
If we initiated a T.38 reINVITE, we would crash if we received any other
1xx response message except 100 if it were followed by a 200 response.
* Made ignore any 1xx response so we do not close out the T.38 negotiation
too early. For good measure we'll now accept any 2xx response as
acceptance of the reINVITE T.38 offer.
res_pjsip_pubsub: Hold module reference for publications.
Incoming publications need to ensure that the module remains
loaded for the lifetime of them. This is now done by holding
a reference to the module while the publication exists. This
mirrors that of inbound subscriptions.
Robert Mordec [Mon, 21 May 2018 12:24:51 +0000 (14:24 +0200)]
app_confbridge: Bridge and announcers not removed if conference ends quickly
If a conference is ended very quickly after it was created (i.e., the
first user immediately hangs up) then the conference bridge and announcer
channels are not removed.
When a conference is created, the push_announcer() function is added to
the playback queue task processor and the conference object reference is
bumped. If a conference is ended while the push_announcer() function is
still going then the ao2_cleanup(conference) at the end of
push_announcer() will call the destructor function -
destroy_conference_bridge().
The destroy_conference_bridge() function will then add the
hangup_playback() task to the playback queue and will wait for it to end.
Since it is already a current task of the playback queue it will wait
forever.
This patch makes the conference thread call push_announcer() directly.
This way the conference object reference bump is not needed. Since the
playback queue task processor is only used by the conference thread
itself, there is no danger of trying to play announcements before the
announcer is pushed to the bridge.
Richard Mudgett [Thu, 28 Jun 2018 17:07:01 +0000 (12:07 -0500)]
AMI SendText action: Fix to use correct thread to send the text.
The AMI action was directly sending the text to the channel driver.
However, this makes two threads attempt to handle media and runs afowl of
CHECK_BLOCKING.
* Queue a read action to make the channel's media handling thread actually
send the text message. This changes the AMI actions success/fail response
to just mean the text was queued to be sent not that the text actually got
sent. The channel driver may not even support sending text messages.
projects / thirdparty / asterisk.git / log
