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Because opus transcoding support cannot be included in the standard Asterisk
distribution, a few codec_opus implementations have popped up. To make it
easier for people to drop in opus support in their own installations, this
patch adds configure checks for libopus.
Review: https://reviewboard.asterisk.org/r/4106/
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gcc on the ARM platform defaults 'char' to 'unsigned char' whereas Intel and
SPARC default to 'signed char'. This is only an issue in the rare cases where
negative values are assigned to a 'char' but this this patch insures
compatibility by detecting platforms that default to 'unsigned' and adding an
'-fsigned-char' flag to _ASTCFLAGS.
If compiling for ARM (native or cross-compile) be sure to run ./bootstrap.sh
and ./configure to regenerate the build files. You shouldn't have to do this
for Intel or SPARC.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4091/
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Fix infinite loop when calling ast_variable_retrieve inside an
ast_category_browse loop when there is more than 1 category with
the same name.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4089/
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This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub
have loaded properly before attempting to load any modules that depend
on them since the module loader system is not currently capable of
resolving module dependencies on its own.
ASTERISK-24312 #close
Reported by: Dafi Ni
Review: https://reviewboard.asterisk.org/r/4062/
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With MALLOC_DEBUG the /main/config config_basic_ops test was causing a
SEGV while doing an ast_category_delete in an ast_category_browse loop.
Apparently this never worked but was also never tested. I removed the
test, added 2 notes to config.h indicating that it's not supported and
added a few lines of code to ast_category_delete to prevent the SEGV
should someone attempt it in the future.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4078/
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Based on feedback from Richard, I created an accessor for
res_phoneprov/ast_phoneprov_std_variable_lookup and added
load priority to AST_MODULE_INFO.
Tested-by: George Joseph
Tested-by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4076/
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This patch provides the capability to manipulate templates and categories
with non-unique names via AMI.
Summary of changes:
GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list
of name_regex=value_regex expressions which will cause only categories whose
variables match all expressions to be considered. The special variable name
TEMPLATES can be used to control whether templates are included. Passing
'include' as the value will include templates along with normal categories.
Passing 'restrict' as the value will restrict the operation to ONLY templates.
Not specifying a TEMPLATES expression results in the current default behavior
which is to not include templates.
UpdateConfig: NewCat now includes options for allowing duplicate category
names, indicating if the category should be created as a template, and
specifying templates the category should inherit from. The rest of the
actions now accept a filter string as defined above. If there are non-unique
category names, you can now update specific ones based on variable values.
To facilitate the new capabilities in manager, corresponding changes had to be
made to config, most notably the addition of filter criteria to many of the
APIs. In some cases it was easy to change the references to use the new
prototype but others would have required touching too many files for this
patch so a wrapper with the original prototype was created. Macros couldn't
be used in this case because it would break binary compatibility with modules
such as res_digium_phone that are linked to real symbols.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4033/
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This patch makes res_phoneprov more modular so other modules (like pjsip)
can provide configuration information instead of res_phoneprov relying solely
on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API
is now exposed which allows config providers to register themselves, set
defaults (server profile, etc) and add user extensions.
* ast_phoneprov_provider_register registers the provider and provides callbacks
for loading default settings and loading users.
* ast_phoneprov_provider_unregister clears the defaults and users.
* ast_phoneprov_add_extension should be called once for each user/extension
by the provider's load_users callback to add them.
* ast_phoneprov_delete_extension deletes one extension.
* ast_phoneprov_delete_extensions deletes all extensions for the provider.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/3970/
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Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.
A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade. With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer. Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.
* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.
* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.
* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.
ASTERISK-24356 #close
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/4034/
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This change gives framehooks a reverse-direction masquerade callback in
addition to chan_fixup_cb similar to the callback added to datastores
to handle the same situation. The new callback provides the same
parameters as the fixup callback, but is called on the new channel's
framehooks before moving framehooks from the old channel to the new
channel. This gives the framehooks an oppurtunity to decide whether
they should remain on the new channel or be removed.
This new callback is used to prevent the PJSIP T.38 framehook from
remaining on a masqueraded channel if the new channel is not also a
PJSIP channel. This was causing a crash when a local channel was
masqueraded into a PJSIP channel and the framehook was executed on the
local channel since the channel's tech private data was not structured
as expected.
Review: https://reviewboard.asterisk.org/r/4001/
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This function acts like strsep with three exceptions...
* The separator is a single character instead of a string.
* Separators inside quotes are treated literally instead of like separators.
* You can elect to have leading and trailing whitespace and quotes
stripped from the result and have '\' sequences unescaped.
Like strsep, ast_strsep maintains no internal state and you can call it
recursively using different separators on the same storage.
Also like strsep, for consistent results, consecutive separators are not
collapsed so you may get an empty string as a valid result.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3989/
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Empty here means that there are no formats in the format_cap structure
or the only format in it is the "none" format.
I've added calls to check the emptiness of a format_cap in a few places
in order to short-circuit operations that would otherwise be pointless
as well as to prevent some assertions from being triggered in cases
where channels with no formats are used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE
arrives.
* It checks that there is a subscription handler for the Event
* It checks that there are body generators for the types in the Accept header
The problem is, there's nothing that ensures that these two things will
actually mesh with each other. For instance, Asterisk will accept a subscription
to MWI that accepts pidf+xml bodies. That doesn't make sense.
With this commit, we add some type information to the mix. Subscription
handlers state they generate data of type X, and body generators state
that they consume data of type X. This way, Asterisk doesn't end up in
some hilariously mismatched situation like the one in the previous paragraph.
ASTERISK-24136 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/3877
Review: https://reviewboard.asterisk.org/r/3878
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When no transport is associated to an endpoint, the AMI output for
PJSIPShowEndpoint indicates an error instead of silently ignoring the
missing transport.
This patch causes the error to appear only if a transport was specified
on the endpoint and the transport doesn't exist. It also fixes an issue
with counting the objects that were actually found.
ASTERISK-24161 #close
ASTERISK-24331 #close
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3998/
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If you call ast_category_insert with a match category that doesn't exist, the
list traverse runs out of 'next' categories and you get a SEGV. This patch
adds check for the end-of-list condition and changes the signature to return
an int for success/failure indication instead of a void.
The only consumer of this function is manager and it was also changed to use
the return value.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3993/
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Changes made during format improvements resulted in the
recording to voicemail option 'm' of the MixMonitor app
writing a zero length duration in the msgXXXX.txt file.
This change introduces a new function ast_ratestream(),
which provides the sample rate of the format associated
with the stream, and updates the app_voicemail function
for ast_app_copy_recording_to_vm to calculate the right
duration.
Review: https://reviewboard.asterisk.org/r/3996/
ASTERISK-24328 #close
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1. The number of file descriptors an ioqueue instance can handle is fixed, so we
now spawn the required number to handle the load.
2. Our transport identifiers were exceeding the range supported by pjnath.
3. The TURN client did not set up client binding causing needless bandwidth usage.
4. The code no longer updates address information on each packet.
5. STUN traffic was getting looped back to Asterisk instead of going through the
TURN server.
6. Synchronization now ensures things are completely setup or destroyed.
7. Logging now reflects the target the TURN server is sending to/receiving from
on our behalf.
ASTERISK-23577 #close
Reported by: Jay Jideliov
ASTERISK-23634 #close
Reported by: Roman Skvirsky
Review: https://reviewboard.asterisk.org/r/3982/
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The context/extension in a CDR is generally considered the destination of a
call. When looking at a 2-party call CDR, users will typically be presented
with the following:
context exten channel dest_channel app data
default 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20
However, if the Dial actually takes place in a Macro, the current behaviour
in 12 will result in the following CDR:
context exten channel dest_channel app data
macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20
The same is true of a GoSub:
context exten channel dest_channel app data
subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20
This generally makes the context/exten fields less than useful.
It isn't hard to preserve these values in the CDR state machine; however, we
need to have something that informs us when a channel is executing a
subroutine. Prior to this patch, there isn't anything that does this.
This patch solves this problem by adding a new channel flag,
AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a
Macro or a GoSub. The CDR engine looks for this value when updating a Party A
snapshot; if the flag is present, we don't override the context/exten on the
main CDR object. In a funny quirk, executing a hangup handler must *not* abide
by this logic, as the endbeforehexten logic assumes that the user wants to see
data that occurs in hangup logic, which includes those subroutines. Since
those execute outside of a typical Dial operation (and will typically have
their own dedicated CDR anyway), this is unlikely to cause any heartburn.
Review: https://reviewboard.asterisk.org/r/3962/
ASTERISK-24254 #close
Reported by: tm1000, Tony Lewis
Tested by: Tony Lewis
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Adds an option to the dial API that marks an outgoing dial as replacing the dialing channel for the purpose of propagating accountcode. When it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on the involved channels with ast_channel_req_accountcodes.
Review: https://reviewboard.asterisk.org/r/3968/
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media has been negotiated.
Testsuite tests will occasionally fail because on reception of a 200 OK SIP response,
an AST_CONTROL_ANSWER frame is queued prior to when media has finished being
negotiated. This is because session supplements are called into before PJSIP's
inv_session code has told us that media has been updated. Sometimes the queued answer
frame is handled by the PBX thread before the ensuing media negotiations occur, causing
a test failure.
As it turns out, there is another place that session supplements could be called into, which is
after media has finished getting negotiated. What this commit introduces is a means for session
supplements to indicate when they wish to be called into when handling an incoming SIP response.
By default, all session supplements will be run at the same point that they were prior to this
commit. However, session supplements may indicate that they wish to be handled earlier than
normal on redirects, or they may indicate they wish to be handled after media has been negotiated.
In this changeset, two session supplements have been updated to indicate a preference for when
they should be run: res_pjsip_diversion executes before handling redirection in order to get
information from the Diversion header, and chan_pjsip now handles responses to INVITEs after
media negotiation to fix the race condition mentioned previously.
ASTERISK-24212 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/3930
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This allows the callerid parsing function to handle malformed input
strings and strings containing escaped and unescaped double quotes.
This also adds a unittest to cover many of the cases where the parsing
algorithm previously failed.
Review: https://reviewboard.asterisk.org/r/3923/
Review: https://reviewboard.asterisk.org/r/3933/
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ASTERISK-24147 #close
Reported by: Edvin Vidmar
Review: https://reviewboard.asterisk.org/r/3908/
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This patch fixes gcc warnings that occur due to the type qualifier 'const'
being ignored on a return type of int.
ASTERISK-24246 #close
Reported by: Shaun Ruffell
patches:
0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch uploaded by Shaun Ruffell (License 5417)
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When a blind transfer occurs that is forced to create a local channel
pair to satisfy the transfer request, information about the local
channel pair is not published. This adds a field to describe that
channel to the blind transfer message struct so that this information
is conveyed properly to consumers of the blind transfer message.
This also fixes a bug in which Stasis() was unable to properly identify
the channel that was replacing an existing Stasis-controlled channel
due to a blind transfer.
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3921/
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If a manager or CLI user attached a mixmonitor to a call running a dynamic
bridge feature while in a bridge, the feature would be interrupted and the
channel would be forcibly kicked out of the bridge (usually ending the call
during a simple 1 to 1 call). This would also occur during any similar action
that could set the unbridge soft hangup flag, so the fix for this was to
remove unbridge from the soft hangup flags and make it a separate thing all
together.
ASTERISK-24027 #close
Reported by: mjordan
Review: https://reviewboard.asterisk.org/r/3900/
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This is to support the backwards compatible changes made in the next version
of Asterisk.
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The patch to catch channels being shoehorned into Stasis() via external
mechanisms also happens to catch Announcer and Recorder channels
because they aren't known to be stasis-controlled channels in the usual
sense. This marks those channels as Stasis()-internal channels and
allows them directly into bridges.
Review: https://reviewboard.asterisk.org/r/3903/
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This patch addresses a few issues:
1) The order of Dial events have been changed when performing a call forward.
The order has now been altered to
1) Dial begins dialing channel A.
2) When A forwards the call to B, we issue the dial end event to channel
A, indicating the dial is being canceled due to a forward to B.
3) When the call to channel B occurs, we then issue a new dial begin to
channel B.
2) Call forwards are now reported on the calling channel, not the peer channel.
3) AMI DialEnd events have been altered to display the extension the call is
being forwarded to when relevant.
4) You can now get the values of channel variables for channels that are not
currently in the Stasis application. This brings the retrieval of channel
variables more in line with the rest of channel read operations since they
may be performed on channels not in Stasis.
ASTERISK-24134 #close
Reported by Matt Jordan
ASTERISK-24138 #close
Reported by Matt Jordan
Patches:
forward-shenanigans.diff uploaded by Matt Jordan (License #6283)
Review: https://reviewboard.asterisk.org/r/3899
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This commit adds the ability for a user to configure
a resource list in pjsip.conf. Subscribing to this
list simultaneously subscribes the subscriber to all
resources listed. This has the potential to reduce
the amount of SIP traffic when loads of subscribers
on a system attempt to subscribe to each others' states.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Fixed the iax.conf bandwidth option. This is the root cause of
ASTERISK-24150.
* Added checks in iax2_request() to ensure that there are actual formats
requested for the new channel to prevent any more fracks from issues like
ASTERISK-24150. This is a consequence of the iax.conf bandwidth option
not working.
* Fixed struct iax2_codec_pref.order member size mismatch issue when
converting to and from the codec preference order list passed over the
wire. In addition the values sent over the wire are now compatible with
previous Asterisk versions.
* Fixed several issues dealing with the struct iax2_codec_pref members.
Off-by-one, array limit errors, and the order/framing members always need
to be updated together.
* Made iax2_request() setup the channel's native format preference order
according to the user's wishes. The new media format strategy needs the
order specified earler.
* Fixed usage of ast_format_compatibility_bitfield2format(). The function
can return NULL if the bitfield was not associated with a function.
* Deleted dead code iax2_codec_pref_getsize() and
iax2_codec_pref_setsize().
* Made iax2_parse_allow_disallow() and iax2_codec_pref_string() call
iax2_codec_pref_to_cap() instead of inlining it.
* Made IAX_CAPABILITY_MEDBANDWIDTH, IAX_CAPABILITY_LOWBANDWIDTH, and
IAX_CAPABILITY_LOWFREE constants again as they were in Asterisk v1.8.
* Renamed prefs to prefs_global so it won't get confused with the local
pref versions.
* Fixed too small buffer in handle_cli_iax2_show_peer().
* Fixed ast_cli() calls in handle_cli_iax2_show_peer() to output complete
lines.
* Changed struct create_addr_info.prefs to be struct iax2_codec_pref as an
optimization so iax2_request() and iax2_call() do less work.
* Fixed a potential deadlock in ast_iax2_new() on an off-nominal path when
the pbx could not get started.
* Made set_config() setup a local prefs list along side the local
capability format bitfield. Once the config is loaded, then the local
copies are put into the global versions.
* Fix unininialized codec_buf in function_iaxpeer().
ASTERISK-24150 #close
Reported by: Scott Griepentrog
Review: https://reviewboard.asterisk.org/r/3890/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This fixes a class of issues where Stasis applications were not made
aware that their channels were being manipulated or replaced by
external entitiessuch as transfers, AMI commands, or dialplan
applications such as Bridge(). Inconsistent information such as
StasisEnd events with unknown channels as a result of masquerades has
also been corrected. To accomplish these fixes, several new fields
were added to blind and attended transfer messages as well as
StasisStart and BridgeAttendedTransfer Stasis events.
ASTERISK-23941 #close
Review: https://reviewboard.asterisk.org/r/3865/
Review: https://reviewboard.asterisk.org/r/3857/
Review: https://reviewboard.asterisk.org/r/3852/
Review: https://reviewboard.asterisk.org/r/3816/
Review: https://reviewboard.asterisk.org/r/3731/
Review: https://reviewboard.asterisk.org/r/3729/
Review: https://reviewboard.asterisk.org/r/3728/
........
Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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state using SIP.
This module uses the inbound and outbound PUBLISH support to exchange device and mailbox
state between Asterisk instances. Each instance is configured to publish to the other and
requires no intermediary server. The functionality provided is similar to the XMPP and
Corosync support.
Review: https://reviewboard.asterisk.org/r/3780/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This module implements the core parts required for doing outbound PUBLISH.
It takes care of configuration, lifetime management, and authentication.
Additional modules implement the specific events that are published.
Review: https://reviewboard.asterisk.org/r/3780/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This introduces stasis.conf and a mechanism to prevent certain message
types from being published. Internally, this works by preventing the
chosen message types from being created which ensures that those
message types can never be published. This patch also adjusts message
publishers such that message payloads are not created if the related
message type is not available.
ASTERISK-23943 #close
Review: https://reviewboard.asterisk.org/r/3823/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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........
r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines
ARI: Add channel technology agnostic out of call text messaging
This patch adds the ability to send and receive text messages from various
technology stacks in Asterisk through ARI. This includes chan_sip (sip),
res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the
endpoints resource, and can be sent directly through that resource, or to a
particular endpoint.
For example, the following would send the message "Hello there" to PJSIP
endpoint alice with a display URI of sip:asterisk@mycooldomain.org:
ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There
This is equivalent to the following as well:
ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There
Both forms are available for message technologies that allow for arbitrary
destinations, such as chan_sip.
Inbound messages can now be received over ARI as well. An ARI application that
subscribes to endpoints will receive messages from those endpoints:
{
"type": "TextMessageReceived",
"timestamp": "2014-07-12T22:53:13.494-0500",
"endpoint": {
"technology": "PJSIP",
"resource": "alice",
"state": "online",
"channel_ids": []
},
"message": {
"from": "\"alice\" <sip:alice@127.0.0.1>",
"to": "pjsip:asterisk@127.0.0.1",
"body": "Watson, come here.",
"variables": []
},
"application": "testsuite"
}
The above was made possible due to some rather major changes in the message
core. This includes (but is not limited to):
- Users of the message API can now register message handlers. A handler has
two callbacks: one to determine if the handler has a destination for the
message, and another to handle it.
- All dialplan functionality of handling a message was moved into a message
handler provided by the message API.
- Messages can now have the technology/endpoint associated with them.
Various other properties are also now more easily accessible.
- A number of ao2 containers that weren't really needed were replaced with
vectors. Iteration over ao2_containers is expensive and pointless when
the lifetime of things is well defined and the number of things is very
small.
res_stasis now has a new file that makes up its structure, messaging. The
messaging functionality implements a message handler, and passes received
messages that match an interested endpoint over to the app for processing.
Note that inadvertently while testing this, I reproduced ASTERISK-23969.
res_pjsip_messaging was incorrectly parsing out the 'to' field, such that
arbitrary SIP URIs mangled the endpoint lookup. This patch includes the
fix for that as well.
Review: https://reviewboard.asterisk.org/r/3726
ASTERISK-23692 #close
Reported by: Matt Jordan
ASTERISK-23969 #close
Reported by: Andrew Nagy
........
r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines
Remove automerge properties :-(
........
r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines
test_message: Fix strict-aliasing compilation issue
........
Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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"module show" .
ASTERISK-23919 #close
Reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/3802
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The previous behavior was to simply set the accountcode of an outgoing
channel to the accountcode of the channel initiating the call. It was
done this way a long time ago to allow the accountcode set on the SIP/100
channel to be propagated to a local channel so the dialplan execution on
the Local;2 channel would have the SIP/100 accountcode available.
SIP/100 -> Local;1/Local;2 -> SIP/200
Propagating the SIP/100 accountcode to the local channels is very useful.
Without any dialplan manipulation, all channels in this call would have
the same accountcode.
Using dialplan, you can set a different accountcode on the SIP/200 channel
either by setting the accountcode on the Local;2 channel or by the Dial
application's b(pre-dial), M(macro) or U(gosub) options, or by the
FollowMe application's b(pre-dial) option, or by the Queue application's
macro or gosub options. Before Asterisk v12, the altered accountcode on
SIP/200 will remain until the local channels optimize out and the
accountcode would change to the SIP/100 accountcode.
Asterisk v1.8 attempted to add peeraccount support but ultimately had to
punt on the support. The peeraccount support was rendered useless because
of how the CDR code needed to unconditionally force the caller's
accountcode onto the peer channel's accountcode. The CEL events were thus
intentionally made to always use the channel's accountcode as the
peeraccount value.
With the arrival of Asterisk v12, the situation has improved somewhat so
peeraccount support can be made to work. Using the indicated example, the
the accountcode values become as follows when the peeraccount is set on
SIP/100 before calling SIP/200:
SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200
acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200
peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100
If a channel already has an accountcode it can only change by the
following explicit user actions:
1) A channel originate method that can specify an accountcode to use.
2) The calling channel propagating its non-empty peeraccount or its
non-empty accountcode if the peeraccount was empty to the outgoing
channel's accountcode before initiating the dial. e.g., Dial and
FollowMe. The exception to this propagation method is Queue. Queue will
only propagate peeraccounts this way only if the outgoing channel does not
have an accountcode.
3) Dialplan using CHANNEL(accountcode).
4) Dialplan using CHANNEL(peeraccount) on the other end of a local
channel pair.
If a channel does not have an accountcode it can get one from the
following places:
1) The channel driver's configuration at channel creation.
2) Explicit user action as already indicated.
3) Entering a basic or stasis-mixing bridge from a peer channel's
peeraccount value.
You can specify the accountcode for an outgoing channel by setting the
CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue
applications. Queue adds the wrinkle that it will not overwrite an
existing accountcode on the outgoing channel with the calling channels
values.
Accountcode and peeraccount values propagate to an outgoing channel before
dialing. Accountcodes also propagate when channels enter or leave a basic
or stasis-mixing bridge. The peeraccount value only makes sense for
mixing bridges with two channels; it is meaningless otherwise.
* Made peeraccount functional by changing accountcode propagation as
described above.
* Fixed CEL extracting the wrong ie value for the peeraccount. This was
done intentionally in Asterisk v1.8 when that version had to punt on
peeraccount.
* Fixed a few places dealing with accountcodes that were reading from
channels without the lock held.
AFS-65 #close
Review: https://reviewboard.asterisk.org/r/3601/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reverting the patch since it was causing a regression and after fixing the
regression, there were no performance gains. At least based on my method
for measurement.
ASTERISK-24050
Review: https://reviewboard.asterisk.org/r/3841/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This flags astobj.h as deprecated, warns people to use astobj2.h instead.
Only netsock.c (also deprecated) still uses astobj.h.
ASTERISK-24069 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3818/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Allow for responses to AMI actions/commands to be documented properly
in XML and displayed via the CLI. Response events are documented
exactly as standard AMI events are documented.
Review: https://reviewboard.asterisk.org/r/3812/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When updating a row, we are currently doing an INSERT OR REPLACE INTO. The
downside to this is that the row is deleted if it exists and then a new row is
inserted. So, we are hitting the disk twice. One for the deletion and one for
the insertion.
This patch changes this statement to an INSERT INTO and if the insert fails
because a row with that key exists, we will IGNORE the failure. Then we will
attempt to perform an UPDATE on the existing row if that row wasn't just
INSERTed.
ASTERISK-24050 #close
Reported by: Michael L. Young
patches:
astdb-insert-update-io-help_trunk_v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3815/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch serves two purposes:
(1) It fixes some bugs with endpoint subscriptions not reporting all of the
channel events
(2) It serves as the preliminary work needed for ASTERISK-23692, which allows
for sending/receiving arbitrary out of call text messages through ARI in a
technology agnostic fashion.
The messaging functionality described on ASTERISK-23692 requires two things:
(1) The ability to send/receive messages associated with an endpoint. This is
relatively straight forwards with the endpoint core in Asterisk now.
(2) The ability to send/receive messages associated with a technology and an
arbitrary technology defined URI. This is less straight forward, as
endpoints are formed from a tech + resource pair. We don't have a
mechanism to note that a technology that *may* have endpoints exists.
This patch provides such a mechanism, and fixes a few bugs along the way.
The first major bug this patch fixes is the forwarding of channel messages
to their respective endpoints. Prior to this patch, there were two problems:
(1) Channel caching messages weren't forwarded. Thus, the endpoints missed
most of the interesting bits (such as channel creation, destruction, state
changes, etc.)
(2) Channels weren't associated with their endpoint until after creation.
This resulted in endpoints missing the channel creation message, which
limited the usefulness of the subscription in the first place (a major use
case being 'tell me when this endpoint has a channel'). Unfortunately,
this meant another parameter to ast_channel_alloc. Since not all channel
technologies support an ast_endpoint, this patch makes such a call
optional and opts for a new function, ast_channel_alloc_with_endpoint.
When endpoints are created, they will implicitly create a technology endpoint
for their technology (if one does not already exist). A technology endpoint is
special in that it has no state, cannot have channels created for it, cannot
be created explicitly, and cannot be destroyed except on shutdown. It does,
however, have all messages from other endpoints in its technology forwarded to
it.
Combined with the bug fixes, we now have Stasis messages being properly
forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar),
where bar has a single channel associated with it and foo has two channels
associated with it. The messages would be forwarded as follows:
channel PJSIP/foo-1 --
\
--> endpoint PJSIP/foo --
/ \
channel PJSIP/foo-2 -- \
---- > endpoint PJSIP
/
channel PJSIP/bar-1 -----> endpoint PJSIP/bar --
ARI, through the applications resource, can:
- subscribe to endpoint:PJSIP/foo and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo
- subscribe to endpoint:PJSIP/bar and get notifications for channels
PJSIP/bar-1 and endpoint PJSIP/bar
- subscribe to endpoint:PJSIP and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar
Note that since endpoint PJSIP never changes, it never has events itself. It
merely provides an aggregation point for all other endpoints in its technology
(which in turn aggregate all channel messages associated with that endpoint).
This patch also adds endpoints to res_xmpp and chan_motif, because the actual
messaging work will need it (messaging without XMPP is just sad).
Review: https://reviewboard.asterisk.org/r/3760/
ASTERISK-23692
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Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Remove functions:
ast_smdi_interface_unref
ast_smdi_md_message_putback
ast_smdi_mwi_message_putback
ast_smdi_md_message destructor
ast_smdi_mwi_message destructor
Includes for astobj.h are removed everywhere it's possible.
ASTERISK-24066 #close
Review: https://reviewboard.asterisk.org/r/3758/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
1. Asterisk was limited in how many formats it could handle.
2. Formats, being a bit field, could not include any attribute information.
A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.
Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.
Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.
Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.
Generally, the new philosophy for handling formats is as follows:
* The ast_format structure is reference counted. This removed a large amount
of the memory allocations and copying that was done in prior versions.
* In order to prevent race conditions while keeping things performant, the
ast_format structure is immutable by convention and lock-free. Violate this
tenet at your peril!
* Because formats are reference counted, codecs are also reference counted.
The Asterisk core generally provides built-in codecs and caches the
ast_format structures created to represent them. Generally, to prevent
inordinate amounts of module reference bumping, codecs and formats can be
added at run-time but cannot be removed.
* All compatibility with the bit field representation of codecs/formats has
been moved to a compatibility API. The primary user of this representation
is chan_iax2, which must continue to maintain its bit-field usage of formats
for interoperability concerns.
* When a format is negotiated with attributes, or when a format cannot be
represented by one of the cached formats, a new format object is created or
cloned from an existing format. That format may have the same codec
underlying it, but is a different format than a version of the format with
different attributes or without attributes.
* While formats are reference counted objects, the reference count maintained
on the format should be manipulated with care. Formats are generally cached
and will persist for the lifetime of Asterisk and do not explicitly need
to have their lifetime modified. An exception to this is when the user of a
format does not know where the format came from *and* the user may outlive
the provider of the format. This occurs, for example, when a format is read
from a channel: the channel may have a format with attributes (hence,
non-cached) and the user of the format may last longer than the channel (if
the reference to the channel is released prior to the format's reference).
For more information on this work, see the API design notes:
https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.
There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).
Reviews:
https://reviewboard.asterisk.org/r/3814
https://reviewboard.asterisk.org/r/3808
https://reviewboard.asterisk.org/r/3805
https://reviewboard.asterisk.org/r/3803
https://reviewboard.asterisk.org/r/3801
https://reviewboard.asterisk.org/r/3798
https://reviewboard.asterisk.org/r/3800
https://reviewboard.asterisk.org/r/3794
https://reviewboard.asterisk.org/r/3793
https://reviewboard.asterisk.org/r/3792
https://reviewboard.asterisk.org/r/3791
https://reviewboard.asterisk.org/r/3790
https://reviewboard.asterisk.org/r/3789
https://reviewboard.asterisk.org/r/3788
https://reviewboard.asterisk.org/r/3787
https://reviewboard.asterisk.org/r/3786
https://reviewboard.asterisk.org/r/3784
https://reviewboard.asterisk.org/r/3783
https://reviewboard.asterisk.org/r/3778
https://reviewboard.asterisk.org/r/3774
https://reviewboard.asterisk.org/r/3775
https://reviewboard.asterisk.org/r/3772
https://reviewboard.asterisk.org/r/3761
https://reviewboard.asterisk.org/r/3754
https://reviewboard.asterisk.org/r/3753
https://reviewboard.asterisk.org/r/3751
https://reviewboard.asterisk.org/r/3750
https://reviewboard.asterisk.org/r/3748
https://reviewboard.asterisk.org/r/3747
https://reviewboard.asterisk.org/r/3746
https://reviewboard.asterisk.org/r/3742
https://reviewboard.asterisk.org/r/3740
https://reviewboard.asterisk.org/r/3739
https://reviewboard.asterisk.org/r/3738
https://reviewboard.asterisk.org/r/3737
https://reviewboard.asterisk.org/r/3736
https://reviewboard.asterisk.org/r/3734
https://reviewboard.asterisk.org/r/3722
https://reviewboard.asterisk.org/r/3713
https://reviewboard.asterisk.org/r/3703
https://reviewboard.asterisk.org/r/3689
https://reviewboard.asterisk.org/r/3687
https://reviewboard.asterisk.org/r/3674
https://reviewboard.asterisk.org/r/3671
https://reviewboard.asterisk.org/r/3667
https://reviewboard.asterisk.org/r/3665
https://reviewboard.asterisk.org/r/3625
https://reviewboard.asterisk.org/r/3602
https://reviewboard.asterisk.org/r/3519
https://reviewboard.asterisk.org/r/3518
https://reviewboard.asterisk.org/r/3516
https://reviewboard.asterisk.org/r/3515
https://reviewboard.asterisk.org/r/3512
https://reviewboard.asterisk.org/r/3506
https://reviewboard.asterisk.org/r/3413
https://reviewboard.asterisk.org/r/3410
https://reviewboard.asterisk.org/r/3387
https://reviewboard.asterisk.org/r/3388
https://reviewboard.asterisk.org/r/3389
https://reviewboard.asterisk.org/r/3390
https://reviewboard.asterisk.org/r/3321
https://reviewboard.asterisk.org/r/3320
https://reviewboard.asterisk.org/r/3319
https://reviewboard.asterisk.org/r/3318
https://reviewboard.asterisk.org/r/3266
https://reviewboard.asterisk.org/r/3265
https://reviewboard.asterisk.org/r/3234
https://reviewboard.asterisk.org/r/3178
ASTERISK-23114 #close
Reported by: mjordan
media_formats_translation_core.diff uploaded by kharwell (License 6464)
rb3506.diff uploaded by mjordan (License 6283)
media_format_app_file.diff uploaded by kharwell (License 6464)
misc-2.diff uploaded by file (License 5000)
chan_mild-3.diff uploaded by file (License 5000)
chan_obscure.diff uploaded by file (License 5000)
jingle.diff uploaded by file (License 5000)
funcs.diff uploaded by file (License 5000)
formats.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
bridges.diff uploaded by file (License 5000)
mf-codecs-2.diff uploaded by file (License 5000)
mf-app_fax.diff uploaded by file (License 5000)
mf-apps-3.diff uploaded by file (License 5000)
media-formats-3.diff uploaded by file (License 5000)
ASTERISK-23715
rb3713.patch uploaded by coreyfarrell (License 5909)
rb3689.patch uploaded by mjordan (License 6283)
ASTERISK-23957
rb3722.patch uploaded by mjordan (License 6283)
mf-attributes-3.diff uploaded by file (License 5000)
ASTERISK-23958
Tested by: jrose
rb3822.patch uploaded by coreyfarrell (License 5909)
rb3800.patch uploaded by jrose (License 6182)
chan_sip.diff uploaded by mjordan (License 6283)
rb3747.patch uploaded by jrose (License 6182)
ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
sip_cleanup.diff uploaded by opticron (License 6273)
chan_sip_caps.diff uploaded by mjordan (License 6283)
rb3751.patch uploaded by coreyfarrell (License 5909)
chan_sip-3.diff uploaded by file (License 5000)
ASTERISK-23960 #close
Tested by: opticron
direct_media.diff uploaded by opticron (License 6273)
pjsip-direct-media.diff uploaded by file (License 5000)
format_cap_remove.diff uploaded by opticron (License 6273)
media_format_fixes.diff uploaded by opticron (License 6273)
chan_pjsip-2.diff uploaded by file (License 5000)
ASTERISK-23966 #close
Tested by: rmudgett
rb3803.patch uploaded by rmudgetti (License 5621)
chan_dahdi.diff uploaded by file (License 5000)
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
rb3814.patch uploaded by rmudgett (License 5621)
moh_cleanup.diff uploaded by opticron (License 6273)
bridge_leak.diff uploaded by opticron (License 6273)
translate.diff uploaded by file (License 5000)
rb3795.patch uploaded by rmudgett (License 5621)
tls_fix.diff uploaded by mjordan (License 6283)
fax-mf-fix-2.diff uploaded by file (License 5000)
rtp_transfer_stuff uploaded by mjordan (License 6283)
rb3787.patch uploaded by rmudgett (License 5621)
media-formats-explicit-translate-format-3.diff uploaded by file (License 5000)
format_cache_case_fix.diff uploaded by opticron (License 6273)
rb3774.patch uploaded by rmudgett (License 5621)
rb3775.patch uploaded by rmudgett (License 5621)
rtp_engine_fix.diff uploaded by opticron (License 6273)
rtp_crash_fix.diff uploaded by opticron (License 6273)
rb3753.patch uploaded by mjordan (License 6283)
rb3750.patch uploaded by mjordan (License 6283)
rb3748.patch uploaded by rmudgett (License 5621)
media_format_fixes.diff uploaded by opticron (License 6273)
rb3740.patch uploaded by mjordan (License 6283)
rb3739.patch uploaded by mjordan (License 6283)
rb3734.patch uploaded by mjordan (License 6283)
rb3689.patch uploaded by mjordan (License 6283)
rb3674.patch uploaded by coreyfarrell (License 5909)
rb3671.patch uploaded by coreyfarrell (License 5909)
rb3667.patch uploaded by coreyfarrell (License 5909)
rb3665.patch uploaded by mjordan (License 6283)
rb3625.patch uploaded by coreyfarrell (License 5909)
rb3602.patch uploaded by coreyfarrell (License 5909)
format_compatibility-2.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch adds a new operation for stored recordings, copy. It takes an
existing stored recording and makes a copy of it in the same directory
or a relative directory under the stored recording directory.
/ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name}
This is particularly useful for voicemail-esque applications, which may need to
copy or move recordings around a directory structure.
Review: https://reviewboard.asterisk.org/r/3768/
ASTERISK-24036 #close
Reported by: Sam Galarneau
Tested by: Sam Galarneau
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Merged revisions 419021 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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If a reference count goes negative, instead of
just logging that fact, be more helpful with a
backtrace and an assert that will DO_CRASH.
This patch also removes the duplicate ao2_bt()
function and cleans up extraneous usage of the
ast_log_backtrace() call.
Review: https://reviewboard.asterisk.org/r/3765/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Whenever possible, audiohooks and framehooks will now be copied over
to the channel that the masquerading channel gets cloned into. This
should occur for all audiohooks and most framehooks. As a result,
in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
deprecated and its behavior is essentially the new default for all
audiohooks, plus some additional audiohooks/framehooks.
Review: https://reviewboard.asterisk.org/r/3721/
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Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Specifically the following equivalents were created:
fax show session -> FAXSession
fax show sessions -> FAXSessions
fax show stats -> FAXStats
Review: https://reviewboard.asterisk.org/r/3666/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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