Matthew Jordan [Sat, 21 Dec 2013 03:35:04 +0000 (03:35 +0000)]
res_pjsip/pjsip_cli: fix compilation error caused by passing ast_free
When wanting to pass *free as a function pointer, ast_free_ptr has to be used
instead of ast_free. This allows it to be compiled with MALLOC_DEBUG enabled.
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Merged revisions 404531 from http://svn.asterisk.org/svn/asterisk/branches/12
David M. Lee [Fri, 20 Dec 2013 22:04:15 +0000 (22:04 +0000)]
ari: Remove support for specifying channel vars during origination.
When we added support for specifying channel variables for an
origination, we didn't consider how that would interact with another
feature, namely specifying request parameters in a JSON request body.
The method of specifying channel variables (as a flat JSON object passed
in the JSON body) interferes with parsing parameters out of the request
body.
Unfortunately, fixing this would be a backward incompatible change. In
the interest of keeping the API sane and keeping our release schedule,
we're dropping the feature for specifying channel variables in the
origination request.
We will bring the feature back soon, as a backward compatible addition
to the API.
Matthew Jordan [Fri, 20 Dec 2013 21:32:13 +0000 (21:32 +0000)]
res_pjsip: Add PJSIP CLI commands
Implements the following cli commands:
pjsip list aors
pjsip list auths
pjsip list channels
pjsip list contacts
pjsip list endpoints
pjsip show aor(s)
pjsip show auth(s)
pjsip show channels
pjsip show endpoint(s)
Also...
Minor modifications made to the AMI command implementations to facilitate
reuse.
New function ast_variable_list_sort added to config.c and config.h to implement
variable list sorting.
(issue ASTERISK-22610)
patches:
pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
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Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12
Mark Michelson [Fri, 20 Dec 2013 20:28:19 +0000 (20:28 +0000)]
Fix issue where PJSIP blind transferer dialog may not complete as planned.
When transferring to a dialplan extension that will not place any outbound
calls, the only control frames that the PJSIP REFER framehook will receive
are inconsequential (such as unhold or srcchange). As such, we shouldn't
allow for the reception of those types of frames prevent us from signaling
to the transferring party that the transfer has completed successfully once
voice frames are read.
Thanks to Jonathan Rose for pointing this out.
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Matthew Jordan [Fri, 20 Dec 2013 20:05:40 +0000 (20:05 +0000)]
res_stasis_device_state: Set resource type for subscriptions to deviceState
The documentation for ARI already specifies that the device state resource when
used for subscribing for events is "deviceState", not "device_state". The code,
however, used "device_state"; although this was inconsistent as well in doxygen
comments in resource_applications.
Because the actual resource being subscribed to is /deviceStates/{device}/, it
makes sense for the resource type specifier to be deviceState.
Note that the key value in the events is still "device_state".
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Matthew Jordan [Fri, 20 Dec 2013 19:25:20 +0000 (19:25 +0000)]
manager: bump version to 2.0.0
AMI has received substantial updates over the past year. Not only has the
syntax been vastly improved and made consistent (which entails many event
changes), but the underlying things that those events convey have changed
substantially as well.
After some conversation in #asterisk-dev, it was agreed that this is a good
time to jump to 2.
At the same time, since ARI will most likely use semantic versioning, we
might as well use that for AMI as well. That also affords us greater meaning
for the AMI version.
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Merged revisions 404421 from http://svn.asterisk.org/svn/asterisk/branches/12
Rusty Newton [Fri, 20 Dec 2013 17:22:27 +0000 (17:22 +0000)]
Documentation: Updates for info about NAT-related settings and fixes for pjsip.conf.sample
Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity.
Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf.
I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options.
(issue ASTERISK-23004)
(closes issue ASTERISK-23004)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3086/
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Joshua Colp [Thu, 19 Dec 2013 18:00:33 +0000 (18:00 +0000)]
res_pjsip: Ignore 401/407 responses for transactions and dialogs we don't know about.
Under normal conditions it is unlikely we will ever receive a response for a transaction
or dialog we don't know about but if any are received ignore them.
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Joshua Colp [Thu, 19 Dec 2013 17:55:28 +0000 (17:55 +0000)]
res_pjsip_session: Fix SDP negotiation when resending an INVITE with authentication.
The process for resending an INVITE with authentication involves restarting the UAC
session. We were incorrectly passing in that a new offer is being sent, causing the
SDP negotiation to get into a (technically speaking) funky state.
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Mark Michelson [Thu, 19 Dec 2013 17:45:21 +0000 (17:45 +0000)]
Fix a deadlock that occurred due to a conflict of masquerades.
For the explanation, here is a copy-paste of the review board explanation:
Initially, it was discovered that performing an attended transfer of a
multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread
started a masquerade and reached the point where it was calling the fixup()
callback on the "original" channel. For chan_pjsip, this involves pushing a
synchronous task to the session's serializer. The problem was that a task ahead
of the fixup task was also attempting to perform a channel masquerade. However,
since masquerades are designed in a way to only allow for one to occur at a
time, the task ahead of the fixup could not continue until the masquerade
already in progress had completed. And of course, the masquerade in progress
could not complete until the task ahead of the fixup task had completed.
Deadlock.
The initial fix was to change the fixup task to be asynchronous. While this
prevented the deadlock from occurring, it had the frightful side effect of
potentially allowing for tasks in the session's serializer to operate on a
zombie channel.
Taking a step back from this particular deadlock, it became clear that the
problem was not really this one particular issue but that masquerades
themselves needed to be addressed. A PJSIP attended transfer operation calls
ast_channel_move(), which attempts to both set up and execute a masquerade. The
problem was that after it had set up the masquerade, the PBX thread had swooped
in and tried to actually perform the masquerade. Looking at changes that had
been made to Asterisk 12, it became clear that there never is any time now that
anyone ever wants to set up a masquerade and allow for the channel thread to
actually perform the masquerade. Everyone always is calling ast_channel_move(),
performs the masquerade itself before returning.
In this patch, I have removed all blocks of code from channel.c that will
attempt to perform a masquerade if ast_channel_masq() returns true. Now, there
is no distinction between setting up a masquerade and performing the
masquerade. It is one operation. The only remaining checks for
ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not
want to interrupt a masquerade by hanging up the channel. Instead, now
ast_hangup() will wait for a masquerade to complete before moving forward with
its operation.
The ast_channel_move() function has been modified to basically in-line the
logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has
been killed off for real. ast_channel_move() now has a lock associated with it
that is used to prevent any simultaneous moves from occurring at once. This
means there is no need to make sure that ast_channel_masq() or
ast_channel_masqr() are already set on a channel when ast_channel_move() is
called. It also means the channel container lock is not pulling double duty by
both keeping the container locked and preventing multiple masquerades from
occurring simultaneously.
The ast_do_masquerade() function has been renamed to do_channel_masquerade()
and is now internal to channel.c. The function now takes explicit arguments of
which channels are involved in the masquerade instead of a single channel.
While it probably is possible to do some further refactoring of this method, I
feel that I would be treading dangerously. Instead, all I did was change some
comments that no longer are true after this changeset.
The other more minor change introduced in this patch is to res_pjsip.c to make
ast_sip_push_task_synchronous() run the task in-place if we are already a SIP
servant thread. This is related to this patch because even when we isolate the
channel masquerade to only running in the SIP servant thread, we would still
deadlock when the fixup() callback is reached since we would essentially be
waiting forever for ourselves to finish before actually running the fixup. This
makes it so the fixup is run without having to push a task into a serializer at
all.
(closes issue ASTERISK-22936)
Reported by Jonathan Rose
res_fax.c: crash on framehook with no dsp in fax detect
In fax_detect_framehook() a null pointer reference can occur where a
voice frame is processed but no dsp is attached to the fax detection
structure. The code block that rejects frames that detection cannot
be processed on is checking for dsp but falls through when it should
instead return, as this change implements.
Richard Mudgett [Thu, 19 Dec 2013 16:52:43 +0000 (16:52 +0000)]
Voicemail: Remove mailbox identifier format (box@context) assumptions in the system.
This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
When Asterisk is shut down, the astdb_atexit() function releases
(finalize) the previously initiated (prepared) SQL statements in
sqlite3. Another thread making a subsequent request can cause a
crash in sqlite3. This patch eliminates that issue by resetting
the statement pointer after it is released/cleared. The sqlite3
code detects the null pointer, and aborts the operation cleanly.
Alexandr Anikin [Thu, 19 Dec 2013 08:35:28 +0000 (08:35 +0000)]
Handle temporary failures on gk registration
Introduce new 'stopped' state for gk client and restart gk client
on failures
Remove ooh323 stack command lock as it is not need now.
(closes issue ASTERISK-21960)
Reported by: Dmitry Melekhov
Patches:
ASTERISK-21960.patch
ASTERISK-21960-stacklockup-2.patch
Tested by: Dmitry Melekhov
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Damien Wedhorn [Thu, 19 Dec 2013 02:59:52 +0000 (02:59 +0000)]
Fixup some skinny bugs causing Fracks and ao2 cleanup issues.
Moved channel locking into setsubstate so that a process can complete
working on a sub before another starts changing it. The existing code
was causing some Fracks with schedule deletion.
Removed multiple rtp cleanup. Now only cleansup up once, fixing ao2
object cleanup issues.
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Matthew Jordan [Thu, 19 Dec 2013 00:50:01 +0000 (00:50 +0000)]
app_cdr,app_forkcdr,func_cdr: Synchronize with engine when manipulating state
When doing the rework of the CDR engine that pushed all of the logic into cdr.c
and made it respond to changes in channel state over Stasis, we knew that
accessing the CDR engine from the dialplan would be "slightly"
non-deterministic. Dialplan threads would be accessing CDRs while Stasis
threads would be updating the state of said CDRs - whereas in the past,
everything happened on the dialplan threads. Tests have shown that "slightly"
is in reality "very".
This patch synchronizes things by making the dialplan applications/functions
that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and
CDR_PROP now all use Stasis to send their requests over to the CDR engine,
and synchronize on the channel Stasis topic via a subscription so that they
return their values/control to the dialplan at the appropriate time.
While going through this, the following changes were also made:
* DISA, which can reset the CDR when a user successfully authenticates, now
just uses the ResetCDR app to do this. This prevents having to duplicate
the same Stasis synchronization logic in that application.
* Answer no longer disables CDRs. It actually didn't work anyway - calling
DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer
time - it just kills all CDRs on that channel, which isn't what the caller
would intend.
Damien Wedhorn [Thu, 19 Dec 2013 00:32:00 +0000 (00:32 +0000)]
Fixup skinny registration following network issues.
On session registration, if device is already reporting that it is
connected to a device, an innocuous packet (update time) is sent to
the already connected device. If the tcp connection is down, the
device will be unregistered and the new connection allowed.
Without this patch, network issues can see a situation where a device
can not reregister until after 3*timeout.
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Merged revisions 404292 from http://svn.asterisk.org/svn/asterisk/branches/12
Kevin Harwell [Wed, 18 Dec 2013 20:33:37 +0000 (20:33 +0000)]
channel locking: Add locking for channel snapshot creation
Original commit message by mmichelson (asterisk 12 r403311):
"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."
The above was initially committed and then reverted at r403398. The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed. Fixed by unreffing the channels.
Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel. Fixed by
unlocking "other->chan"
(closes issue ASTERISK-22709)
Reported by: John Bigelow
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Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12
Alexandr Anikin [Wed, 18 Dec 2013 19:36:39 +0000 (19:36 +0000)]
Introduce new config option 'aniasdni'. If yes then asterisk set dialed number as own id back to the caller
on incoming h.323 calls. Option can be set globally or per user section.
(closes issue ASTERISK-22020)
Reported by: Ross Beer
Joshua Colp [Wed, 18 Dec 2013 19:28:05 +0000 (19:28 +0000)]
channels: Return allocated channels locked.
This change makes ast_channel_alloc return allocated channels
locked. By doing so no other thread can acquire, lock, and manipulate
the channel before it is completely set up.
Jonathan Rose [Tue, 17 Dec 2013 23:57:52 +0000 (23:57 +0000)]
tests: fix ast_bridge_base_new calls not using the additional arguments
r404042 gave ast_bridge_base_new two new arguments for setting a bridge creator
and name. Unfortunately since a couple test modules aren't compiled by default,
I missed the fact that this change impacted those tests and caused compilation
failures against them.
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Merged revisions 404048 from http://svn.asterisk.org/svn/asterisk/branches/12
Jonathan Rose [Tue, 17 Dec 2013 23:25:49 +0000 (23:25 +0000)]
bridging: Give bridges a name and a known creator
Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.
Joshua Colp [Tue, 17 Dec 2013 18:26:09 +0000 (18:26 +0000)]
framehooks: Re-iterate if framehook provides different frame.
Framehooks can be used in a reactive manner to execute specific logic
when a frame is received with a certain type and payload. Since it is
possible for framehooks to provide frames it was possible for this
reactive framehook to be unaware of frames it is looking for.
This change makes it so that when framehooks return a modified frame
the code will now re-iterate (from the beginning) and call any
previous framehooks that have not provided a modified frame themselves.
app_voicemail: Voicemail callback registration/unregistration function improvements.
* The voicemail registration/unregistration functions now take a struct of
callbacks instead of a lengthy parameter list of callbacks.
* The voicemail registration/unregistration functions now prevent a
competing module from interfering with an already registered callback
supplying module.
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r403780 | rmudgett | 2013-12-13 13:55:31 -0600 (Fri, 13 Dec 2013) | 8 lines
test_voicemail_api: Add check for a registered voicemail provider before tests.
It is much nicer diagnosing a test failure if app_voicemail is actually
loaded.
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Merged revisions 403726 from http://svn.asterisk.org/svn/asterisk/trunk
Matthew Jordan [Tue, 17 Dec 2013 12:59:49 +0000 (12:59 +0000)]
ari/resource_channels: When creating a channel, specify a default format (SLIN)
When creating channels via ARI, the current code fails to provide any default
format capabilities. For non-virtual channels this isn't really a problem -
the channels typically receive their capabilities as a result of the
underlying channel driver configuration. For virtual channels (such as Local
channels), the lack of any format capabilities causes the Asterisk core to
make some 'odd' choices with respect to the translation paths. The issue
reporter had some paths that had 3 hops on each channel leg, causing multiple
transcodings and some really crappy audio/performance.
By specifying a baseline of SLIN, we prevent that from occurring. Note that
this is what AMI does when it performs an Originate, as does res_clioriginate.
Review: https://reviewboard.asterisk.org/r/3068/
(issue ASTERISK-22962)
Reported by: Matt DiMeo
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Merged revisions 403993 from http://svn.asterisk.org/svn/asterisk/branches/12
David M. Lee [Mon, 16 Dec 2013 19:11:51 +0000 (19:11 +0000)]
security: Inhibit execution of privilege escalating functions
This patch allows individual dialplan functions to be marked as
'dangerous', to inhibit their execution from external sources.
A 'dangerous' function is one which results in a privilege escalation.
For example, if one were to read the channel variable SHELL(rm -rf /)
Bad Things(TM) could happen; even if the external source has only read
permissions.
Execution from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of asterisk.conf.
Although doing so is not recommended.
Also, the ABI was changed to something more reasonable, since Asterisk
12 does not yet have a public release.
Jonathan Rose [Mon, 16 Dec 2013 18:31:12 +0000 (18:31 +0000)]
transfers: Fix bug setting both BLINDTRANSFER and ATTENDEDTRANSFER
The ast_bridge_set_transfer_variables function is supposed to wipe whichever
variable isn't being set. Instead it was setting both to the new value. Oops.
(issue AFS-24)
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Merged revisions 403957 from http://svn.asterisk.org/svn/asterisk/branches/12
pbx.c: put copy of ast_exten.data on stack to prevent memory corruption
During dialplan execution in pbx_extension_helper(), the contexts global
read lock prevents link list corruption, but was released with a pointer
to the ast_exten and data later used in variable substitution. Instead,
this patch removes pbx_substitute_variables() and locates a copy of the
ast_exten data on the stack before releasing the lock, where ast_exten
could get free'd by another thread performing a module reload.
(issue AST-1179)
Reported by: Thomas Arimont
(issue AST-1246)
Reported by: Alexander Hömig
Review: https://reviewboard.asterisk.org/r/3055/
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app_sms: BufferOverflow when receiving odd length 16 bit message
This patch prevents an infinite loop overwriting memory when
a message is received into the unpacksms16() function, where
the length of the message is an odd number of bytes.
(closes issue ASTERISK-22590)
Reported by: Jan Juergens
Tested by: Jan Juergens
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Merged revisions 403856 from http://svn.asterisk.org/svn/asterisk/branches/12
Joshua Colp [Sat, 14 Dec 2013 17:28:21 +0000 (17:28 +0000)]
res_pjsip: Apply outbound proxy to all SIP requests.
Objects which are involved in SIP request creation and sending
now allow an outbound proxy to be specified. For cases where
an endpoint is used the outbound proxy specified there will
be applied.
(closes issue ASTERISK-22673)
Reported by: Antti Yrjola
Joshua Colp [Sat, 14 Dec 2013 17:19:41 +0000 (17:19 +0000)]
res_stasis: Expose event for call forwarding and follow forwarded channel.
This change adds an event for when an originated call is redirected to
another target. This event contains the original channel and the newly
created channel. If a stasis subscription exists on the original originated
channel for a stasis application then a new subscription will also be
created on the stasis application to the redirected channel. This allows
the application to follow the call path completely.
Jonathan Rose [Fri, 13 Dec 2013 20:13:22 +0000 (20:13 +0000)]
Transfers: Make Asterisk set ATTENDEDTRANSFER/BLINDTRANSFER more reliably
There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER
wouldn't be set on channels involved with blind and attended transfers.
This would happen with features that were initialized by channel driver
specific mechanisms in multiparty calls. This patch resolves those cases
while attempted to keep the behavior for setting those variables as
consistent as possible.
Kevin Harwell [Fri, 13 Dec 2013 18:33:25 +0000 (18:33 +0000)]
bridge_native_rtp: Deadlock during 4-way conference creation
The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
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Merged revisions 403767 from http://svn.asterisk.org/svn/asterisk/branches/12
Kevin Harwell [Fri, 13 Dec 2013 17:19:23 +0000 (17:19 +0000)]
ARI: Allow specifying channel variables during a POST /channels
Added the ability to specify channel variables when creating/originating a
channel in ARI. The variables are sent in the body of the request and should
be formatted as a single level JSON object. No nested objects allowed.
For example: {"variable1": "foo", "variable2": "bar"}.
(closes issue ASTERISK-22872)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3052/
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Kevin Harwell [Fri, 13 Dec 2013 16:38:57 +0000 (16:38 +0000)]
ARI: Adding a channel to a bridge while a live recording is active blocks
Added the ability to have rules that are checked when adding and/or removing
channels to/from a bridge. In this case, if a channel is currently recording
and someone attempts to add it to a bridge an "is recording" rule is checked,
fails, and a 409 conflict is returned.
Also command functions now return an integer value that can be descriptive of
what kind of problems, if any, occurred before or during execution.
realtime: Create extensions in alembic ast-db-manage contribution
When the alembic scripts were written for creating Asterisk
realtime databases the extensions table for dialplan wasn't
included. This update creates the extensions table.
(closes issue ASTERISK-22815)
Reported by: Zone Conkle
Review: https://reviewboard.asterisk.org/r/3064/
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Merged revisions 403713 from http://svn.asterisk.org/svn/asterisk/branches/12
Jonathan Rose [Thu, 12 Dec 2013 19:18:06 +0000 (19:18 +0000)]
chan_pjsip: Revert r403587
This patch was intended to eliminate a deadlock that occurs when
masquerades occur in pjsip channels, but has some potential side
effects. Mark Michelson is currently working on addressing this
problem from another angle.
(issue ASTERISK-22936)
Reported by: Jonathan Rose
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Merged revisions 403705 from http://svn.asterisk.org/svn/asterisk/branches/12
Kevin Harwell [Wed, 11 Dec 2013 20:24:50 +0000 (20:24 +0000)]
res_pjsip_messaging: send message to a default outbound endpoint
In some cases messages need to be sent to a direct URI (sip:<ip address>). This
patch adds in that support by using a default outbound endpoint. When sending
messages, if no endpoint can be found then the default one is used.
To facilitate this a new default_outbound_endpoint option was added to the
globals section for pjsip.conf.
Russell Bryant [Wed, 11 Dec 2013 19:22:05 +0000 (19:22 +0000)]
Reset peer outboundproxy on sip.conf reload
If you set a peer's outboundproxy and then removed it from the config,
this would not get picked up in a config reload. This patch fixes that
by resetting it in set_peer_defaults().
Richard Mudgett [Wed, 11 Dec 2013 19:19:24 +0000 (19:19 +0000)]
app_voicemail: Voicemail callback registration/unregistration function improvements.
* The voicemail registration/unregistration functions now take a struct of
callbacks instead of a lengthy parameter list of callbacks.
* The voicemail registration/unregistration functions now prevent a
competing module from interfering with an already registered callback
supplying module.
Matthew Jordan [Wed, 11 Dec 2013 13:06:30 +0000 (13:06 +0000)]
func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip
This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
* RTP information, including source/destination media addresses, whether or
not the media is secure, held, and other properties.
* RTCP information. This includes sets of parseable information, as well as
individual statistic attriutes.
* PJSIP information. This includes URIs, local/remote signalling addresses,
whether or not the signalling is secure, and other properties.
* The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
function to obtain more detailed endpoint information.
Matthew Jordan [Wed, 11 Dec 2013 12:31:57 +0000 (12:31 +0000)]
func_pjsip_endpoint: Add PJSIP_ENDPOINT function for querying endpoint details
This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query,
for any endpoint, any property configured on an endpoint. This function is a
companion to the CHANNEL function, which can be used to extract the endpoint
name for a channel.
Mark Michelson [Tue, 10 Dec 2013 15:15:13 +0000 (15:15 +0000)]
Fix correct authentication behavior for artificial endpoint.
When switching to using a vector for authentication, I initialized
the vector for the artificial endpoint to be of size 1. However, this
does not result in AST_VECTOR_SIZE() returning 1 since there isn't
actually anything in the vector.
Rather than trifle with the vector by putting unnecessary elements in,
I simply changed the callback in res_pjsip_authenticator_digest.c to
explicitly report that the artificial endpoint requires authentication.
Richard Mudgett [Mon, 9 Dec 2013 19:24:58 +0000 (19:24 +0000)]
Reverting regex part of -r403545 at request of file.
res_sorcery_astdb.c: Fix get multiple records by regex.
* Fix sorcery_astdb_retrieve_regex() pattern matching. Let the regexec()
function match the stored key values instead of having astdb prefilter
them. Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.
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Richard Mudgett [Mon, 9 Dec 2013 18:50:20 +0000 (18:50 +0000)]
res_sorcery_astdb.c: Fix get multiple records by regex.
* Fix sorcery_astdb_retrieve_regex() pattern matching. Let the regexec()
function match the stored key values instead of having astdb prefilter
them. Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.
* Fix off nominal memory leak in sorcery_astdb_retrieve_regex().
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Richard Mudgett [Mon, 9 Dec 2013 18:32:57 +0000 (18:32 +0000)]
sorcery: Eliminate shadowing a varaible that caused confusion.
* Eliminated shadowing of the __ast_sorcery_apply_config() name parameter
causing confusion.
* Fix potential crash from sorcery.conf user input in
__ast_sorcery_apply_config() if the user supplied a malformed config line
that is missing the sorcery object type name.
* Remove redundant test in __ast_sorcery_apply_config(). !config and
config == CONFIGS_STATUS_FILEMISSING are identical.
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Joshua Colp [Mon, 9 Dec 2013 18:32:02 +0000 (18:32 +0000)]
endpoints: Keep a reference to channel ids when creating snapshot.
The snapshot process for endpoints uses the channel ids present
on the endpoint itself. Without keeping a reference it was possible
for the strings to be freed underneath any consumer of an endpoint
snapshot.
A reference is now held by the snapshot to the channel ids and
released when the snapshot is destroyed.
(issue ASTERISK-22801)
Reported by: Matt Jordan
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Mark Michelson [Mon, 9 Dec 2013 16:42:59 +0000 (16:42 +0000)]
Create function for retrieving Mixmonitor instance data.
For the time, this is only useful for retrieving the filename.
The purpose of this function is to better facilitate multiple
mixmonitors per channel. Setting a MIXMONITOR_FILENAME channel
variable is not conducive to such behavior, so allowing finer
grained access to individual mixmonitor properties improves
the situation. The MIXMONITOR_FILENAME channel variable is still
set, though, so there is no worry about backwards compatibility.
Joshua Colp [Mon, 9 Dec 2013 16:41:43 +0000 (16:41 +0000)]
res_pjsip_nat: Add NAT module to session dialogs.
Due to the way pjproject internally works it was possible for the
NAT module to not be invoked on messages with-in a session dialog.
This means that the various parts of the message would not get rewritten
with the source IP address and port.
This change uses a session supplement to add the NAT module
to the dialog on the first incoming or outgoing INVITE.
Mark Michelson [Mon, 9 Dec 2013 16:10:05 +0000 (16:10 +0000)]
Switch PJSIP auth to use a vector.
Since Asterisk has a vector API now, places where arrays are manually
resized don't really make sense any more. Since the auth work in PJSIP
was freshly-written, it was easy to reform it to use a vector.
Matthew Jordan [Mon, 9 Dec 2013 03:21:56 +0000 (03:21 +0000)]
res_fax_spandsp: Always init T.38 session to avoid crashes during state change
Prior to this patch, res_fax_spandsp was conservative with how it initialized
the spandsp T.38 context. It would only initialize it if the driver thought
the current state was a T.38 fax. While this works fine in nominal situations,
in certain off nominal situations, res_fax_spandsp can believe that a T.38
fax will not occur when in fact one has started. In particular, this was
discovered when res_fax would fall back to audio after timing out on a T.38
upgrade. The SIP channel driver would continue to retry the re-INVITE and -
if the remote end responded after res_fax timed out with a 200 OK - a T.38
frame would be delivered to the res_fax stack when it no longer expected it.
As it turns out, there does not appear to be any downside to always
initializing the T.38 context, other than the actual memory allocation.
Since that avoids this off nominal situation (and others which are equally
likely hard to predict), this is the safest way to avoid this problem.
Much thanks to Torrey as well for providing a scenario that reproduces this
issue.
Matthew Jordan [Sun, 8 Dec 2013 05:59:46 +0000 (05:59 +0000)]
res_config_sqlite: Check for CDR unregistration failures
If the CDR unregistration fails due to an inflight CDR, the
res_config_sqlite module needs to bail on unloading itself. Otherwise,
the config could be unloaded (including the CDR table name) while the
CDR engine posts a CDR to the still registered backend, resulting in
a crash.
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David M. Lee [Thu, 5 Dec 2013 22:09:20 +0000 (22:09 +0000)]
ari: Fix deadlock problem with functions that use autoservice.
The code for getting channel variables from ARI assumed that you needed
to lock the channel in order to properly execute functions and read
channel variables. Apparently, this is not the case, since any dialplan
function that puts the channel into autoservice deadlocks when
attempting to remove the channel from autoservice.
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Joshua Colp [Wed, 4 Dec 2013 20:54:52 +0000 (20:54 +0000)]
res_pjsip_t38: Don't pass T.38 control frames through to other hooks.
This crept up during gateway testing where the gateway would receive
the request to negotiate and assume it came from the remote side, causing
the gateway state machine to go a little, to a use a technical term,
"wonky".
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Mark Michelson [Wed, 4 Dec 2013 18:41:01 +0000 (18:41 +0000)]
Initialize the hash value argument to pj_hash_get() to 0.
Passing a non-zero value causes PJLIB to use the given input as the
hash value. Passing zero causes the parameter to become an output parameter
that receives the hash value that was computed based on the given key.
This change essentially makes ast_sip_dict_get() properly retrieve the
desired value.
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Joshua Colp [Tue, 3 Dec 2013 18:01:36 +0000 (18:01 +0000)]
res_pjsip_session: Add support for PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag.
Newer versions of PJSIP have changed to using a flag for the
PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a
configure check to detect the presence of the flag and use it if found.
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Richard Mudgett [Tue, 3 Dec 2013 17:35:54 +0000 (17:35 +0000)]
sorcery, bucket: Change observer remove calls to take const callbacks struct.
* Make ast_sorcery_observer_remove() accept a const callbacks struct.
* Make ast_sorcery_observer_remove() tolerant of the sorcery parameter
being NULL. Now it can be called within a module unload routine if the
sorcery initialization fails.
* Fix ast_sorcery_observer_add() to fail if the container link fails.
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Mark Michelson [Tue, 3 Dec 2013 17:07:29 +0000 (17:07 +0000)]
Add channel locking for channel snapshot creation.
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
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Joshua Colp [Tue, 3 Dec 2013 16:39:13 +0000 (16:39 +0000)]
media_index: Make media indexing tolerable of bad symlinks.
Media indexing will now skip over files and directories that stat
will not return information about. This can occur under normal
conditions when a symbolic link points to a location that no longer
exists.
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Merged revisions 403312 from http://svn.asterisk.org/svn/asterisk/branches/12
Alexandr Anikin [Mon, 2 Dec 2013 18:12:57 +0000 (18:12 +0000)]
Check and reject non-digits e164 values on peers and general sections in ooh323.conf
Regenerate e164 endpoint list on reload ooh323
(issue ASTERISK-22901)
Reported by: Cyril CONSTANTIN
Patches:
ASTERISK-22901.patch
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Joshua Colp [Sun, 1 Dec 2013 20:04:55 +0000 (20:04 +0000)]
res_pjsip_t38: Add the framehook to the channel only on first INVITE.
The check for determining whether the T.38 framehook should be added to
the channel or not has now been changed to guarantee adding only occurs
on the first incoming or outgoing INVITE.
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Merged revisions 403258 from http://svn.asterisk.org/svn/asterisk/branches/12
Joshua Colp [Sun, 1 Dec 2013 19:58:08 +0000 (19:58 +0000)]
res_pjsip_transport_websocket: Fix security events and simplify implementation.
Transport type determination for security events has been simplified to use
the type present on the message itself instead of searching through configured
transports to find the transport used.
The actual WebSocket transport has also been simplified. It now leverages the
existing PJSIP transport manager for finding the active WebSocket transport
for outgoing messages. This removes the need for res_pjsip_transport_websocket
to store a mapping itself.
(closes issue ASTERISK-22897)
Reported by: Max E. Reyes Vera J.
Joshua Colp [Thu, 28 Nov 2013 02:12:45 +0000 (02:12 +0000)]
res_pjsip_sdp_rtp: Don't produce an invalid media stream with no formats.
Depending on configuration it was possible for a media stream to be
created without any media formats. The produced SDP would fail internal
validation and cause a crash.
The code will now no longer add media streams with no formats to the SDP,
allowing it to pass validation and work.
(closes issue ASTERISK-22858)
Reported by: Anthony Messina
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