Age | Commit message (Collapse) | Author |
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Most callers of ast_channel_make_compatible() happen before the channels
enter a two party bridge. With the new bridging framework, two party
bridging technologies may also call ast_channel_make_compatible() when
there is more than one thread involved with the two channels.
* Added channel lock protection in set_format() and
ast_channel_make_compatible_helper() when dealing with the channel's
native formats while setting up a translation path.
* Fixed best_src_fmt and best_dst_fmt usage consistency in
ast_channel_make_compatible_helper(). The call to
ast_translator_best_choice() got them backwards.
* Updated some callers of ast_channel_make_compatible() and the function
documentation. There is actually a difference between the two channels
passed in.
* Fixed the deadlock potential in res_fax.c dealing with
ast_channel_make_compatible(). The deadlock potential was already there
anyway because res_fax called ast_channel_make_compatible() with chan
locked.
(closes issue ASTERISK-22542)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2915/
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(closes issue ASTERISK-22631)
Reported by: Kevin Harwell
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Drop an error log message to debug level 1 since distributed device
state functions correctly when receiving this message and it spams the
logs.
(closes issue ASTERISK-22410)
Reported by: abelbeck
Patches:
asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
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parameter.
(closes issue ASTERISK-22680)
Reported by: John Bigelow
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This makes it clear that the ARI API calls for listing channels and
bridges will list all channels or bridges in the system and not just
those that are in or are controlled by a Stasis application.
(closes issue ASTERISK-22635)
Reported by: Kevin Harwell
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Review: https://reviewboard.asterisk.org/r/2911/
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This adds the list of expected errors to the /bridges/{bridgeId}/record
ARI documentation so that outbound 4xx errors validate properly.
Previously, this would result in a response validation failure.
(closes issue ASTERISK-22627)
Reported by: Joshua Colp
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This change updates how we handle toggle events, rather then create two
different function names, we'll just use POST / DELETE from HTTP to handle it.
Review: https://reviewboard.asterisk.org/r/2906/
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If the server didn't exist that we are trying to register to the log message
would say that a 408 was received from that server when in reality one wasn't.
Added log messages stating no response was received if the response does not
exist.
(closes issue ASTERISK-22554)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2893/
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The module info block was repeated twice. Once is sufficient.
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The INVITE session state callback wrongly assumes that a session will always exist, but
when rapidly terminating the session this assumption goes out the window. As all handler
code for the INVITE session state callback requires the session it will now just exit
immediately if no session exists.
(closes issue ASTERISK-22668)
Reported by: John Bigelow
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When generating the list of authentication credentials to pass to
PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not
always NULL-terminated. This sometimes resulted in incorrect text for
the realm and a failure to match the realm for authentication purposes
which was causing the outbound nominal auth pjsip basic call test to
bounce. This now uses the pj_str_t that contains the realm instead of
generating a new one. Thanks to John Bigelow for helping to narrow this
down.
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r400508 | dlee | 2013-10-03 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line
Corrected response class for stopPlayback
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r400842 | dlee | 2013-10-10 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line
Correct some ARI wiki rendering errors
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r400843 | dlee | 2013-10-10 14:26:19 -0500 (Thu, 10 Oct 2013) | 1 line
Updated /play resource docs. The playback of http: resources isn't implemented... yet
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r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5 lines
Fix a stupid copy/paste error in ARI docs.
Patches:
ari-doc-patch.txt uploaded by jbigelow (license 5091)
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This change fixes two issues when setting an outbound proxy:
1. The outbound proxy URI was not parsed and validated during configuration.
2. If an outgoing dialog was created and the outbound proxy could not be set an assertion would
occur because the usage count on the dialog was not decremented.
The documentation has also been updated to specify that a full URI must be specified for
the outbound proxy.
(closes issue ASTERISK-22672)
Reported by: Antti Yrjola
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Using 'lu' will produce a compiler warning for some versions of gcc and on some
architectures. 'z' should be portable as a format specifier for size_t.
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This patch adds support to the PJSIP stack in Asterisk for SIP header
manipulation. Note that this is analagous to SIPAddHeader/SIPRemoveHeader.
For PJSIP_HEADER, an incoming supplemental session callback is registered that
takes the pjsip_hdrs from the incoming session and stores them in a linked
list in the session datastore. Calls to PJSIP_HEADER traverse over the list
and return the nth matching header where 'n' is the 'number' argument to the
function.
When adding a header, the first call creates a datastore and linked list and
adds the datastore to the session. The header is then created as a pjsip_hdr
and added to the list. An outgoing supplemental session callback then
traverses the list and adds the headers to the outgoing pjsip_msg.
When removing a header, the list created with PJSIP_HEADER(add,...) is
traversed and all matching entries are removed.
(closes issue ASTERISK-22498)
Reported by: George Joseph
patch:
res_pjsip_header_funcs_v1.patch uploaded by george.joseph (License 6322)
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pjsip_strerror is only aware of PJSIP-specific error
codes. pj_strerror() is aware of all PJProject error
codes and OS-specific error codes.
This specifically fixes an oft-seen error in transport
configuration code where EADDRINUSE would result in
"Unknown PJSIP error 120098" instead of a useful
message.
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Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4
candidates are also added.
(closes issue ASTERISK-21917)
Reported by: Torrey Searle
Patches:
0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334)
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If you run Asterisk in the background and then connect to
it through a separate console, the thread that runs CLI commands
is not registered with PJLIB. Thus PJLIB does not like it when
you attempt to send OPTIONS requests from that thread. So now
we push the task into the threadpool, which we know to be registered
with PJLIB.
Thanks to Antti Yrjola for reporting this.
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The https://reviewboard.asterisk.org/r/2888/ review changes manager to not
subscribe to stasis when it is disabled for performance reasons. When
manager is disabled app_queue and res_agi decline to load and fail to
clean up what they have already allocated.
* Made app_queue and res_agi clean up allocated resources when they
decline to load.
* Made app_queue and res_agi use their own subscriptions to the stasis
topics instead of borrowing manager's message router structure
inappropriately.
(closes issue ASTERISK-22604)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/2902/
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This patch makes the res_pjsip_logger do a few things... First, it
will be built and installed by default now, so end users won't need
to enable it in menuselect. Second, while it is loaded, it no longer
will immediately issue log messages. Upon loading, it is in the
disabled state and must be turned on with the new CLI command. The
CLI command 'pjsip set logger <on/off/host> has been added and can be
used to do the following:
pjsip set logger on:
Enables logger for all PJSIP traffic
pjsip set logger off:
Disables logger for all PJSIP traffic
pjsip set logger host <host>:
Enables logger for the specific host
Review: https://reviewboard.asterisk.org/r/2900/
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This patch adds an /applications API to ARI, allowing explicit management of
Stasis applications.
* GET /applications - list current applications
* GET /applications/{applicationName} - get details of a specific application
* POST /applications/{applicationName}/subscription - explicitly subscribe to
a channel, bridge or endpoint
* DELETE /applications/{applicationName}/subscription - explicitly unsubscribe
from a channel, bridge or endpoint
Subscriptions work by a reference counting mechanism: if you subscript to an
event source X number of times, you must unsubscribe X number of times to stop
receiveing events for that event source.
Review: https://reviewboard.asterisk.org/r/2862
(issue ASTERISK-22451)
Reported by: Matt Jordan
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specified.
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external media address.
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specifying a direction.
(closes issue ASTERISK-22637)
Reported by: Scott Griepentrog
Patch by Matt Jordan, whose office I have taken over in the name of Canada.
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This fixes a bug where the SSRC field on multicast RTP can be stuck at
0 which can cause problems for endpoints trying to make sense of
incoming streams.
(closes issue ASTERISK-22567)
Reported by: Simone Camporeale
Patches:
22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536)
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Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.
Review: https://reviewboard.asterisk.org/r/2879
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and external_media_address is set.
The callback function for changing the media address in streams wrongly assumes that a connection line
will always be present. This is false as no line is present if a stream has been rejected.
(closes issue ASTERISK-22645)
Reported by: Rusty Newton
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r400318 | mmichelson | 2013-10-02 17:08:49 -0500 (Wed, 02 Oct 2013) | 12 lines
Remove unnecessary waits from stasis.
Since caches are updated on publisher threads, there is no need
to wait for the cache updates to occur after a stasis message
is published.
In the case of chan_pjsip device state changes, this set of
changes caused an improvement to performance.
Review: https://reviewboard.asterisk.org/r/2890
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r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed, 02 Oct 2013) | 3 lines
Remove svn:mergeinfo property.
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will always have a channel.
When starting up or shutting down this assumption is false.
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* There were several places in ARI where an external library was mallocing
memory that must always be released with free(). When MALLOC_DEBUG is
enabled, free() is redirected to the MALLOC_DEBUG version. Since the
external library call still uses the normal malloc(), MALLOC_DEBUG
complains that the freed memory block is not registered and will not free
it. These cases must use ast_std_free().
* Changed calls to asprintf() and vasprintf() to the equivalent
ast_asprintf() and ast_vasprintf() versions respectively.
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Due to the asynchronous design of the PJMEDIA SDP negotiator it was possible for
the SDP to be negotiated *after* a channel was created and after it was being wait
on by an application. It is only after negotiation occurs that the file descriptors
for RTP are placed on the channel. Since the channel was already being waited on
these file descriptors were not monitored, causing incoming media to never be read.
This change wakes up any application waiting on the channel so that added file
descriptors end up being monitored.
(closes issue AST-1227)
Reported by: John Bigelow
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operation.
(issue ASTERISK-22625)
Reported by: Scott Griepentrog
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potentially initialize it at the same time.
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r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
Minor performance bump by not allocate manager variable struct if we don't need it
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r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
Stasis performance improvements
This patch addresses several performance problems that were found in
the initial performance testing of Asterisk 12.
The Stasis dispatch object was allocated as an AO2 object, even though
it has a very confined lifecycle. This was replaced with a straight
ast_malloc().
The Stasis message router was spending an inordinate amount of time
searching hash tables. In this case, most of our routers had 6 or
fewer routes in them to begin with. This was replaced with an array
that's searched linearly for the route.
We more heavily rely on AO2 objects in Asterisk 12, and the memset()
in ao2_ref() actually became noticeable on the profile. This was
#ifdef'ed to only run when AO2_DEBUG was enabled.
After being misled by an erroneous comment in taskprocessor.c during
profiling, the wrong comment was removed.
Review: https://reviewboard.asterisk.org/r/2873/
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r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
Taskprocessor optimization; switch Stasis to use taskprocessors
This patch optimizes taskprocessor to use a semaphore for signaling,
which the OS can do a better job at managing contention and waiting
that we can with a mutex and condition.
The taskprocessor execution was also slightly optimized to reduce the
number of locks taken.
The only observable difference in the taskprocessor implementation is
that when the final reference to the taskprocessor goes away, it will
execute all tasks to completion instead of discarding the unexecuted
tasks.
For systems where unnamed semaphores are not supported, a really
simple semaphore implementation is provided. (Which gives identical
performance as the original taskprocessor implementation).
The way we ended up implementing Stasis caused the threadpool to be a
burden instead of a boost to performance. This was switched to just
use taskprocessors directly for subscriptions.
Review: https://reviewboard.asterisk.org/r/2881/
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r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
Optimize how Stasis forwards are dispatched
This patch optimizes how forwards are dispatched in Stasis.
Originally, forwards were dispatched as subscriptions that are invoked
on the publishing thread. This did not account for the vast number of
forwards we would end up having in the system, and the amount of work it
would take to walk though the forward subscriptions.
This patch modifies Stasis so that rather than walking the tree of
forwards on every dispatch, when forwards and subscriptions are changed,
the subscriber list for every topic in the tree is changed.
This has a couple of benefits. First, this reduces the workload of
dispatching messages. It also reduces contention when dispatching to
different topics that happen to forward to the same aggregation topic
(as happens with all of the channel, bridge and endpoint topics).
Since forwards are no longer subscriptions, the bulk of this patch is
simply changing stasis_subscription objects to stasis_forward objects
(which, admittedly, I should have done in the first place.)
Since this required me to yet again put in a growing array, I finally
abstracted that out into a set of ast_vector macros in
asterisk/vector.h.
Review: https://reviewboard.asterisk.org/r/2883/
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r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
Remove dispatch object allocation from Stasis publishing
While looking for areas for performance improvement, I realized that an
unused feature in Stasis was negatively impacting performance.
When a message is sent to a subscriber, a dispatch object is allocated
for the dispatch, containing the topic the message was published to, the
subscriber the message is being sent to, and the message itself.
The topic is actually unused by any subscriber in Asterisk today. And
the subscriber is associated with the taskprocessor the message is being
dispatched to.
First, this patch removes the unused topic parameter from Stasis
subscription callbacks.
Second, this patch introduces the concept of taskprocessor local data,
data that may be set on a taskprocessor and provided along with the data
pointer when a task is pushed using the ast_taskprocessor_push_local()
call. This allows the task to have both data specific to that
taskprocessor, in addition to data specific to that invocation.
With those two changes, the dispatch object can be removed completely,
and the message is simply refcounted and sent directly to the
taskprocessor.
Review: https://reviewboard.asterisk.org/r/2884/
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This adds a global option in chan_sip to allow it to continue
attempting registration if a 403 is received, clearing the cached nonce
and treating it as a non-fatal response. Normally, this would cause
registration attempts to that endpoint to stop.
This also adds a similar per-outbound-registration option to chan_pjsip
which allows the retry interval to be altered for 403 responses to
REGISTER requests.
(closes issue ASTERISK-17138)
Review: https://reviewboard.asterisk.org/r/2874/
Reported by: Rudi
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We forgot to add documentation for res_pjsip_notify, which would prevent it
from being loaded. Whoops.
This patch also updates res_pjsip_notify to use pjsip_notify.conf, which now
has its own sample file in the configs directory as well.
Review: https://reviewboard.asterisk.org/r/2835/
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RTCP's calculation of the number of lost packets in an RTP stream is based on
that stream's sequence number count, the number of received packets, and how
many packets we expect to receive. When the SSRC for an RTP stream changes,
there can - and almost always will be - a large jump in the next packet's
timestamp and sequence number. If we don't reset the number of received
packets, sequence number count, and other metrics used by RTCP, the next RR/SR
report will use the previous SSRC's values to calculate the lost packet count
for the new SSRC - resulting in a very large number of lost packets.
This patch modifies res_rtp_asterisk such that, if it detects a SSRC change, it
will reset the various values used by the RTCP calculations. From the
perspective of RTCP, this appears as a new media stream - which is what it is.
Review: https://reviewboard.asterisk.org/r/2886/
(closes issue AST-1174)
Reported by: Thomas Arimont
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* app_cdr left the ResetCDR application registered.
* res_parking leaked a ref to config global.
(closes issue ASTERISK-22566)
Reported by: Corey Farrell
Patches:
ASTERISK-22566-r2.patch (license #5909) patch uploaded by Corey Farrell
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There was a collision of mod_data use on the transaction between using a nat
hook and an session response callback. During state change it was assumed
what was in the mod_data was nothing or the response callback. However, it
was possible for it to also contain a nat hook thus resulting in a bad cast
and a crash.
Added the ability to store multiple data elements in mod_data via a hash table.
In this instance, mod_data now stores a hash table of the two values that can
be retrieved using an associated string key.
(closes issue ASTERISK-22394)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2843/
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While handling a registration request a race condition could occur if/when two+
clients registered at the same time. This happened when one request obtained a
copy of the current contacts for an AOR and another request did the same before
the first request updated. Thus the second would update and overwrite the first
(or vice-versa depending on which actually updated first). In the case of it
being the same contact two "add" events would be raised.
pjsip registration handling is now serialized to alleviate this issue.
(closes issue AST-1213)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2860/
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tags inside a <note>
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Endpoints and Transports
Re-using some of Mark Michelson's text from an E-mail discussion for:
* Modifying synopsis for both options
* Adding description to both options
* Changing name of "external_media_address" for Endpoint configuration to "media_address" in anticipation of the option name being changed. (As it is not really specific to external destinations)
(issue ASTERISK-22405)
(closes issue ASTERISK-22405)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2850/
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res_pjsip and res_pjsip_session.
During load time in res_pjsip if an error occurred the operation would attempt to rollback all
operations done during load. This is not permitted by PJSIP as it will assert if the operation has
not been done. This fix changes the code so it will only rollback what has been initialized already.
Further changes also prevent res_pjsip and res_pjsip_session from being unloaded. This is due to
limitations within PJSIP itself. The library environment can only be changed to a certain extent
and does not provide the ability, currently, to deinitialize certain required functionality.
(closes issue ASTERISK-22474)
Reported by: Corey Farrell
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Moved rtcp_report RAII_VAR declaration into the loop so it is unref'ed
after every loop. Moved message_blob to loop and switched it to a regular
variable. The regular variable was used since message_blob is used in a
very contained way.
(closes issue ASTERISK-22565)
Reported by: Corey Farrell
Patches:
rtcp_report-leak.patch (license #5909) patch uploaded by Corey Farrell
Tested by: Corey Farrell
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pjsip's message technology was being registered as 'sip', which was causing it
to not load due it conflicting with chan_sip's registered 'sip' technology for
messaging. It now registers as 'pjsip'. However, due to this change the "to"
field for outgoing pjsip messages need to be prefixed with 'pjsip:' instead of
'sip:'. Incoming messages to res_pjsip_messaging will automatically have their
"to" fields altered in order to accommodate the change. Outgoing messages also
handle changing it back to 'sip' before being sent so the pjsip library will
properly handle it.
(closes issue ASTERISK-22445)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2833/
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The endpoint option does not apply to communication with external entities. Rather,
the option is applied to all communications with the endpoint. The external_media_address
transport configuration option may override the endpoint option if it turns out that
we are going to be communicating with an external entity.
Two things of note:
1) I have not updated the XML documentation. This is being taken care of by Rusty as part
of his work on issue ASTERISK-22405
2) This commit is likely to cause testsuite failures since there are tests that use the
external_media_address endpoint option, and they will need to be changed over. Well, I'm
planning to get that updated ASAP after this commit.
(closes issue ASTERISK-22528)
reported by Rusty Newton
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Fixes regression introduced by -r374096.
* Made res_speech.export.in export ast_* symbols instead of specific
functions.
* Made app_speech_utils.c declare that it is dependent upon res_speech.
(issue ASTERISK-17136)
Reported by: Richard Kenner
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The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.
* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.
* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.
* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.
* Made all callers of ast_bridge_impart() check the return value. It is
important. As a precaution, I also made the compiler complain now if it
is not checked.
* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.
An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.
(closes issue ASTERISK-22072)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2845/
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