Mark Michelson [Wed, 20 Aug 2014 20:04:43 +0000 (20:04 +0000)]
Move evaluation of set_var options in pjsip to the end of channel initialization.
This allows for set_var to override certain defaults such as caller ID and codec
values. This also fixes a test suite regression. The "set_var" test suite test attempted
to use set_var to override caller ID, but a recent change caused that to no longer work.
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Kinsey Moore [Wed, 20 Aug 2014 13:06:33 +0000 (13:06 +0000)]
Stasis: Add information to blind transfer event
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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Kinsey Moore [Wed, 20 Aug 2014 12:39:39 +0000 (12:39 +0000)]
AMI: Add AllVariables parameter to Status
This adds the AllVariables parameter to the Status AMI action such that
if defined and set to "true", all channel variables will be reported in
the subsequent Status event(s). This parameter does not negate the
functionality of the "Variables" parameter so that global variables and
dialplan functions can be requested.
Jonathan Rose [Tue, 19 Aug 2014 16:36:30 +0000 (16:36 +0000)]
ARI: Fix a bug where /channels/{channelID}/continue doesn't execute PBX
If /channels/{channelID}/continue is called on a channel that was originated
without a PBX (such as the ARI command POST channel with a stasis application
argument), the channel will not start dialplan execution. This patch will now
run the PBX out of the stasis execution if the channel doesn't currently have
an active PBX upon continuing.
Richard Mudgett [Tue, 19 Aug 2014 16:16:03 +0000 (16:16 +0000)]
chan_pjsip: Fix attended transfer connected line name update.
A calls B
B answers
B SIP attended transfers to C
C answers, B and C can see each other's connected line information
B completes the transfer
A has number but no name connected line information about C
while C has the full information about A
I examined the incoming and outgoing party id information handling of
chan_pjsip and found several issues:
* Fixed ast_sip_session_create_outgoing() not setting up the configured
endpoint id as the new channel's caller id. This is why party A got
default connected line information.
* Made update_initial_connected_line() use the channel's CALLERID(id)
information. The core, app_dial, or predial routine may have filled in or
changed the endpoint caller id information.
* Fixed chan_pjsip_new() not setting the full party id information
available on the caller id and ANI party id. This includes the configured
callerid_tag string and other party id fields.
* Fixed accessing channel party id information without the channel lock
held.
* Fixed using the effective connected line id without doing a deep copy
outside of holding the channel lock. Shallow copy string pointers can
become stale if the channel lock is not held.
* Made queue_connected_line_update() also update the channel's
CALLERID(id) information. Moving the channel to another bridge would need
the information there for the new bridge peer.
* Fixed off nominal memory leak in update_incoming_connected_line().
* Added pjsip.conf callerid_tag string to party id information from
enabled trust_inbound endpoint in caller_id_incoming_request().
George Joseph [Mon, 18 Aug 2014 20:20:59 +0000 (20:20 +0000)]
func_config: Change 'Not Found' message from ERROR to DEBUG
When you call the CONFIG dialplan function with the name of a variable that
doesn't exist in the target context you get an ERROR. This does nothing but
clutter up the logs with messages that may be perfectly acceptable. Just
because a variable wasn't in the context doesn't mean it's an error. Maybei
t's optional or just needs to be defaulted or ignored.
This patch changes the log level from ERROR to DEBUG. If a dialplan developer
wants to debug their dialplan they still canby setting the console debug level
as needed.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3919/
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res/ari/resource_channels: Don't return allocation failure on failed function
If a function fails to execute, it is most likely due to one of two reasons:
(1) The function doesn't exist or can't be read from
(2) The function is dangerous and is restricted based on the user's permissions
Currently we return allocation failure, which is incorrect. This updates the
reason code to more accurately reflect why the request failed.
Matthew Jordan [Mon, 18 Aug 2014 00:57:01 +0000 (00:57 +0000)]
Improve call forwarding reporting, especially with regards to ARI.
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)
Matthew Jordan [Sun, 17 Aug 2014 23:29:34 +0000 (23:29 +0000)]
apps/app_meetme: Fix crash when publishing MeetMe messages with no channel
The same function, meetme_stasis_generate_msg, handles creating and publishing
Stasis message both when there are channels in the MeetMe conference and when
there are no channels in the conference. When the performance improvement was
made to use cached snapshots, this created a situation where Asterisk would
crash: obtaining a cached snapshot is not NULL tolerant.
This patch restores the previous implementation, which used a NULL safe set
of routines to produce a blob containing the channel snapshot (if available)
and information about the MeetMe conference.
Matthew Jordan [Sun, 17 Aug 2014 23:10:21 +0000 (23:10 +0000)]
apps/app_dial: Fix Dial 'z' option
The 'z' option is supposed to disable the dial timeout in the case of a call
forward. Unfortunately, the wrong timeout timer was passed to the do_forward
function, resulting in the option not working.
Matthew Jordan [Sun, 17 Aug 2014 22:35:27 +0000 (22:35 +0000)]
configure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc
Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is
executed with optimization. This "help" unfortunately results in re-definition
warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This
patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning.
Review: https://reviewboard.asterisk.org/r/3912/
ASTERISK-24032 #close
Reported by: Kilburn
Tested by: Kilburn, wdoekes
patches:
1.8.diff uploaded by cloos (License 5956)
10.diff uploaded by cloos (License 5956)
11.diff uploaded by cloos (License 5956)
12.diff uploaded by cloos (License 5956)
13.diff uploaded by cloos (License 5956)
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Jonathan Rose [Fri, 15 Aug 2014 17:26:12 +0000 (17:26 +0000)]
Bridging: Fix a behavioral change when checking if a channel is leaving a bridge
r420934 introduced some failures in the test suite. Upon investigating, it was
discovered that differences in the way we were evaluating whether a channel was in
the process of leaving a bridge were causing some reinvites not to occur (mostly
reinvites back to Asterisk when ending a call). This patch fixes that behavioral
change.
ASTERISK-24027 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3910/
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Matthew Jordan [Fri, 15 Aug 2014 15:50:46 +0000 (15:50 +0000)]
app_voicemail/app: Remove test events that were duplicated by r421059
Moving the test event raised when a file is played back (which occurred in
r421059) broke the ever loving snot out of the voicemail tests. This caused
duplicate test events to get raised, as app_voicemail and main/app were raising
events prior to call ast_streamfile. The voicemail tests did not enjoy getting
multiple events.
Since raising the playback event in ast_streamfile is far more useful to the
vast majority of tests, this patch keeps the call there and simply removes the
extraneous calls that duplicated the event.
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Matthew Jordan [Thu, 14 Aug 2014 21:16:32 +0000 (21:16 +0000)]
res/res_hep_rtcp: Remove dependency on PJSIP
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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Matthew Jordan [Thu, 14 Aug 2014 19:21:51 +0000 (19:21 +0000)]
cel: Make sure channels in extra fields include their unique IDs as well
CEL typically tracks a lot of information using the unique ID of the channel.
This is typically needed due to tying events together using the linked ID of
the various channels involved in a "call", which is derived from the channel ID
of the oldest channel involved in a bridge (or in the case of a Dial, the
parent channel).
Previously, we had updated the extra fields to include the involved channel
names, but forgot to put in the unique ID. This patch corrects that error.
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Kinsey Moore [Wed, 13 Aug 2014 16:56:14 +0000 (16:56 +0000)]
PJSIP: Prevent crash no-URI contacts
This prevents a crash from occurring when a contact with no URI is used
for the creation of an outbound out-of-dialog request with no
associated endpoint.
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Jonathan Rose [Wed, 13 Aug 2014 16:24:37 +0000 (16:24 +0000)]
Bridges: Fix feature interruption/unintended kick caused by external actions
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.
Richard Mudgett [Tue, 12 Aug 2014 23:36:37 +0000 (23:36 +0000)]
res_stasis_snoop.c: Fix off nominial exit path leaving Snoop channel locked and not hungup.
* Made use ast_copy_string() instead of strcpy() for snoop uniqueid for
safety. There is no guarantee that the max channel uniqueid length will
remain the same as the snoop uniqueid space.
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Kinsey Moore [Mon, 11 Aug 2014 18:38:15 +0000 (18:38 +0000)]
Stasis: Allow internal channels directly into bridges
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.
Mark Michelson [Mon, 11 Aug 2014 17:40:07 +0000 (17:40 +0000)]
Fix crashing unit tests with regards to RLS.
The unit tests require a sorcery.conf file that has been
set up to store resource lists in memory rather than retrieving
from configuration.
With a setup that is not conducive to running the tests, a fault
in sorcery currently causes Asterisk to crash when attempting to
run any of the tests.
To get around the crash, this adds a function that verifies the
current environment and marks the tests as "not run" if the setup
is not correct.
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Matthew Jordan [Mon, 11 Aug 2014 13:57:53 +0000 (13:57 +0000)]
res_hep: Remove disabling of modules
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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Walter Doekes [Mon, 11 Aug 2014 10:41:07 +0000 (10:41 +0000)]
general: Fix memory Corruption in __ast_string_field_ptr_build_va.
If the space left in a stringfield is between 0 and
(alignof(ast_string_field_allocation)-1) adding new data would cause
memory corruption, because we would assume enough space (unsigned
underrun).
Thanks Arnd Schmitter for reporting and finding out the cause!
Matthew Jordan [Mon, 11 Aug 2014 00:14:53 +0000 (00:14 +0000)]
app_queue: Add RealTime support for queue rules
This patch gives the optional ability to keep queue rules in RealTime. It is
important to note that with this patch:
(a) Queue rules in RealTime are only examined on module load/reload
(b) Queue rules are loaded both from the queuerules.conf file as well as the
RealTime backend
To inform app_queue to examine RealTime for queue rules, a new setting has been
added to queuerules.conf's general section "realtime_rules". RealTime queue
rules will only be used when this setting is set to "yes".
The schema for the database table supports a rule_name, time, min_penalty, and
max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or
'+' literal is provided. Otherwise, the penalties are treated as constants.
Rule: default
- After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust
QUEUE_MIN_PENALTY to 20
Rule: test2
- After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust
QUEUE_MIN_PENALTY to 30
- After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust
QUEUE_MIN_PENALTY by -11
- After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
QUEUE_MIN_PENALTY to 112
Rule: test3
- After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust
QUEUE_MIN_PENALTY to 4564
Rule: test_rule
- After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust
QUEUE_MIN_PENALTY to 15
If you use RealTime, the queue rules will be always reloaded on a module
reload, even if the underlying file did not change. With the option disabled,
the rules will only be reloaded if the file was modified.
Review: https://reviewboard.asterisk.org/r/3607/
ASTERISK-23823 #close
Reported by: Michael K
patches:
app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621)
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Kinsey Moore [Thu, 7 Aug 2014 20:24:15 +0000 (20:24 +0000)]
Stasis: Ensure transfer messages follow validation rules
This makes Stasis() event generation for transfer messages follow
validation rules. Currently, ast_json_null() is being used in place of
omitting a key entirely which falls afoul of these validation rules.
https://reviewboard.asterisk.org/r/3892/
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Mark Michelson [Thu, 7 Aug 2014 19:26:32 +0000 (19:26 +0000)]
Add support for RFC 4662 resource list subscriptions.
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.
Richard Mudgett [Thu, 7 Aug 2014 18:51:16 +0000 (18:51 +0000)]
chan_iax2: Several media format fixes.
* 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
Kinsey Moore [Thu, 7 Aug 2014 15:30:19 +0000 (15:30 +0000)]
Stasis: Convey transfer information to applications
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.
Joshua Colp [Thu, 7 Aug 2014 14:37:26 +0000 (14:37 +0000)]
res_pjsip_publish_asterisk: Add support for exchanging device and mailbox 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.
Joshua Colp [Thu, 7 Aug 2014 14:35:09 +0000 (14:35 +0000)]
res_pjsip_outbound_publish: Add module which provides outbound PUBLISH support.
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.
Matthew Jordan [Thu, 7 Aug 2014 14:17:54 +0000 (14:17 +0000)]
pbx: Filter out pattern matching hints in responses sent to ExtensionStateList
Hints that are a pattern match are technically stored in the hint container in
the same fashion as concrete implementations of hints. The pattern matching
hints, however, are not "real" in the sense that things can subscribe to them:
rather, they are stored in the hints container so that when a subscription is
made a "real" hint can be generated for the subscription if one does not yet
exist. The extension state core takes care of this correctly by matching
against non-pattern matching extensions prior to pattern matching extensions.
Because of this, however, the ExtensionStateList AMI action was returning
pattern matching hints when executed. These hints are meaningless from the
perspective of AMI clients: their state will never change, they cannot be
subscribed to, and events would never normally be generated from them. As such,
we now filter these out of the response.
Matthew Jordan [Thu, 7 Aug 2014 03:04:49 +0000 (03:04 +0000)]
build_tools: Skip managerEvent combining for AMI action responses
AMI action responses can (and will) reference AMI events that they return.
These event references and definitions should not be combined with AMI events
raised elsewhere in the code, as they are specifically tied to the AMI action
that raised them.
Richard Mudgett [Wed, 6 Aug 2014 18:12:48 +0000 (18:12 +0000)]
Fix alembic script to work properly in offline mode.
When run in offline mode, this would attempt to check the database for
the presence of a type it was going to try to create. I now check the
context to see if we're running in offline mode and change a parameter
accordingly.
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George Joseph [Wed, 6 Aug 2014 16:12:26 +0000 (16:12 +0000)]
pbx_lua: fix regression with global sym export and context clash by pbx_config.
ASTERISK-23818 (lua contexts being overwritten by contexts of the same name in
pbx_config) surfaced because pbx_lua, having the AST_MODFLAG_GLOBAL_SYMBOLS
set, was always force loaded before pbx_config. Since I couldn't find any
reason for pbx_lua to export it's symbols to the rest of Asterisk, I simply
changed the flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't
realize was that the symbols need to be exported not because Asterisk needs
them but because any external Lua modules like luasql.mysql need the base
Lua language APIs exported (ASTERISK-17279).
Back to ASTERISK-23818... It looks like there's an issue in pbx.c where
context_merge was only merging includes, switches and ignore patterns if
the context was already existing AND has extensions, or if the context was
brand new. If pbx_lua is loaded before pbx_config, the context will exist
BUT pbx_lua, being implemented as a switch, will never place extensions in
it, just the switch statement. The result is that when pbx_config loads,
it never merges the switch statement created by pbx_lua into the final
context.
This patch sets pbx_lua's modflag back to AST_MODFLAG_GLOBAL_SYMBOLS and adds
an "else if" in context_merge that catches the case where an existing context
has includes, switchs or ingore patterns but no actual extensions.
ASTERISK-23818 #close
Reported by: Dennis Guse
Reported by: Timo Teräs
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3891/
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Kinsey Moore [Wed, 6 Aug 2014 12:55:28 +0000 (12:55 +0000)]
Stasis: Allow message types to be blocked
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.
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:
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
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r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines
Rusty Newton [Mon, 4 Aug 2014 19:47:06 +0000 (19:47 +0000)]
Manager - Improve documentation for manager commands Getvar and Setvar.
The documentation for these commands did not make it clear that they could
accept expressions and functions. Modified to make this clear, but tried
not to be overly explicit.
ASTERISK-21178 #close
Reported by: Rusty Newton
Tested by: Rusty Newton
Kinsey Moore [Sat, 2 Aug 2014 03:37:25 +0000 (03:37 +0000)]
Manager: Add PJSIPShowEndpoint[s] documentation
This adds a large swath of response documentation for PJSIPShowEndpoint
and PJSIPShowEndpoints AMI commands. It relies heavily on the existing
text in the configInfo documentation via xi:include tags to avoid
documentation duplication.
Mark Michelson [Fri, 1 Aug 2014 14:48:35 +0000 (14:48 +0000)]
Add ContactStatusDetail to PJSIPShowEndpoint AMI output.
Now when running PJSIPShowEndpoint, you will receive a
ContactStatusDetail for each bound contact that Asterisk
is qualifying. This information includes the URI of the
contact, current reachability, and roundtrip time.
Matthew Jordan [Thu, 31 Jul 2014 11:57:51 +0000 (11:57 +0000)]
res_hep_rtcp: Add module that sends RTCP information to a Homer Server
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
........
Merged revisions 419823 from http://svn.asterisk.org/svn/asterisk/branches/12
Matthew Jordan [Thu, 31 Jul 2014 11:49:40 +0000 (11:49 +0000)]
xmldocs: Add support for an <example> tag in the Asterisk XML Documentation
This patch adds support for an <example /> tag in the XML documentation schema.
For CLI help, this doesn't change the formatting too much:
- Preceeding white space is removed
- Unlike with para elements, new lines are preserved
However, having an <example /> tag in the XML schema allows for the wiki
documentation generation script to surround the documentation with {code} or
{noformat} tags, generating much better content for the wiki - and allowing us
to put dialplan examples (and other code snippets, if desired) into the
documentation for an application/function/AMI command/etc.
Kinsey Moore [Wed, 30 Jul 2014 18:32:25 +0000 (18:32 +0000)]
manager: Add state list commands
This patch adds three new AMI commands:
* ExtensionStateList (pbx.c) - list all known extension state hints
and their current statuses. Events emitted by the list action are
equivalent to the ExtensionStatus events.
* PresenceStateList (res_manager_presencestate) - list all known
presence state values. Events emitted are generated by the stasis
message type, and hence are PresenceStateChange events.
* DeviceStateList (res_manager_devicestate) - list all known device
state values. Events emitted are generated by the stasis message
type, and hence are DeviceStateChange events.
Patch-by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3799/
res_pjsip_session: Fix race condition where redirecting information may not be set.
Since the PJSIP INVITE session module is invoked before any session supplements it was
possible for it to handle a redirect before the res_pjsip_diversion module interpreted
and set redirecting information on the channel. This would cause the redirecting
information to get lost.
This patch ensures that session supplements are *always* invoked before a redirect occurs
by explicitly calling them in the redirect handler.
res_pjsip_pidf_body_generator / res_pjsip_xpidf_body_generator: Ensure local entity is unquoted.
The local entity as provided by PJSIP is quoted within '<' and '>'. As a result placing
this value into XML will result in malformed XML being produced. This patch now unquotes
the local entity so it can go safely into the XML.
loader: Fix an infinite loop when printing modules using "module show".
When creating the alphabetical sorted list each module is added to a list
temporarily. On the second iteration each module already has a pointer to
another module, causing stuff to go into a loop.
ARI: report duration values in LiveRecording objects
This patch adds three new fields to the LiveRecording model:
- total_duration: the total length of the live recording
- talking_duration: optional. The duration of talking energy that was
detected while the recording was made.
- silence_duration: optional. The duration of silence that was detected while
the recording was made.
These values are reported in the RecordingFinished ARI event.
When a DSP is enabled on the channel during the recording - which occurs when
the recording is created with max_silence_seconds (indicating that the user
actually cares about how much silence is in the file), we will report the
talking_duration and silence_duration in addition to the total_duration.
Matthew Jordan [Fri, 25 Jul 2014 14:27:52 +0000 (14:27 +0000)]
module loader: Unload modules in reverse order of their start order
When Asterisk starts a module (calling its load_module function), it re-orders
the module list, sorting it alphabetically. Ostensibly, this was done so that
the output of 'module show' listed modules in alphabetic order. This had the
unfortunate side effect of making modules with complex usage patterns
unloadable. A module that has a large number of modules that depend on it is
typically abandoned during the unloading process. This results in its memory
not being reclaimed during exit.
Generally, this isn't harmful - when the process is destroyed, the operating
system will reclaim all memory allocated by the process. Prior to Asterisk 12,
we also didn't have many modules with complex dependencies. However, with
the advent of ARI and PJSIP, this can make make unloading those modules
successfully nearly impossible, and thus tracking memory leaks or ref debug
leaks a real pain.
While this patch is not a complete overhaul of the module loader - such an
effort would be beyond the scope of what could be done for Asterisk 13 -
this does make some marginal improvements to the loader such that modules
like res_pjsip or res_stasis *may* be made properly un-loadable in the future.
1. The linked list of modules has been replaced with a doubly linked list. This
allows traversal of the module list to occur backwards. The module shutdown
routine now walks the global list backwards when it attempts to unload
modules.
2. The alphabetic reorganization of the module list on startup has been
removed. Instead, a started module is placed at the end of the module list.
3. The ast_update_module_list function - which is used by the CLI to display
the modules - now does the sorting alphabetically itself. It creates its own
linked list and inserts the modules into it in alphabetic order. This allows
for the intent of the previous code to be maintained.
This patch also contains a fix for res_calendar. Without calendar.conf, the
calendar modules were improperly bumping the use count of res_calendar, then
failing to load themselves. This patch makes it so that we detect whether or
not calendaring is enabled before altering the use count.
app_bridgewait: Remove possibility of race condition between channels leaving/joining.
Bridges created by app_bridgewait previously had the "dissolve when empty" flag set.
This caused the bridge core to destroy them when the last channel had left. This
introduced a race condition where we may have a reference to the bridge but it is
not actually joinable when we try to join it. This flag has now been removed and the
bridge is guaranteed to be joinable at all times.
bridge: Make "bridge destroy" only available in developer mode and add "all" to "bridge kick".
The "bridge destroy" CLI command is invasive to bridges and can leave them in an unexpected
state for the users of them. Since this command may be useful for developers it is now
only available when developer mode is available. To take its place "all" has been added
as a valid option to the "bridge kick" CLI command. It will kick all of the channels
in the bridge out.
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:
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.
Michael L. Young [Thu, 24 Jul 2014 21:01:37 +0000 (21:01 +0000)]
core/db: Revert Patch Added In Attempt To Improve I/O Performance
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.
........
chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy
sip_subscribe_mwi_destroy calls sip_destroy on the reference counted
mwi->call. This results in the fields of mwi->call being freed, but
mwi->call itself it leaked. If other code is still using mwi->call
it can cause problems. This change uses dialog_unref instead, to
balance the ref provided by sip_alloc().
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