Richard Mudgett [Thu, 22 Jan 2015 19:30:12 +0000 (19:30 +0000)]
Bridge core: Pass a ref with the swap channel when joining a bridge.
When code imparts a channel into a bridge to swap with another channel, a
ref needs to be held on the swap channel to ensure that it cannot
dissapear before finding it in the bridge.
* The ast_bridge_join() swap channel parameter now always steals a ref for
the swap channel. This is the only change to the bridge framework's
public API semantics.
* bridge_channel_internal_join() now requires the bridge_channel->swap
channel to pass in a ref.
stasis transfer: fix a race condition on stasis bridge push
After a bridge transfer completes where a local replacement
channel is used, a stasis transfer message with the details
of the transfer is sent. This is processed by stasis which
then sets the stasis app name and replaced channel snapshot
on the replacement channel.
However, since a separate thread was already started to run
stasis on the new replacement channel, a race was on to see
if the message processing would be completed before the app
name was needed, otherwise the channel would be hung up.
This change moves the calls used to set the stasis app name
and the replace snapshot to the bridge_stasis_push function
callback from the bridge transfer logic, allowing the steps
to be completed earlier and more deterministically, and the
race elimianted.
NOTE: the swap channel parameter to bridge_stasis_push (and
thus all bridge push callbacks) must always be present when
performing a swap with another channel.
ASTERISK-24649 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4341/
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Matthew Jordan [Thu, 22 Jan 2015 14:23:41 +0000 (14:23 +0000)]
apps/app_voicemail: Trigger MWI notification with MixMonitor m() option
The MixMonitor m() option allows a recording to be pushed to a specific
voicemail mailbox. If the message is delivered to the mailbox's INBOX, however,
no MWI notification is currently raised.
This patch corrects the issue by properly calling notify_new_state from the
msg_create_from_file function. This will cause MWI to be triggered if the
message was placed in the mailbox's INBOX.
Matthew Jordan [Wed, 21 Jan 2015 13:36:52 +0000 (13:36 +0000)]
channels/chan_sip: Fix registration leak during reload
When the SIP registrations were migrated to using ao2 in what was then trunk,
the explicit destruction of the registrations on module reload was removed and
not replaced with an ao2 equivalent. Debugging done by Stefan Engström, the
issue reporter, on ASTERISK-24673 confirmed that the reference in the
registry_list container was being leaked.
Since the purpose of cleanup_all_regs is to prep a registration for
destruction, this function now calls an ao2_callback function callback with the
OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the registrations.
This cleans up each registration, and also removes it from the registration
container registry_list.
Review: https://reviewboard.asterisk.org/r/4355/
ASTERISK-24640 #close
Reported by: Max Man
ASTERISK-24673 #close
Reported by: Stefan Engström
Tested by: Stefan Engström
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Matthew Jordan [Wed, 21 Jan 2015 13:27:55 +0000 (13:27 +0000)]
AMI: Add documentation for the missing Cdr/CEL events.
This patch adds AMI event documentation for the Cdr and CEL AMI events.
Note that while these events do share fields with each other and with other
channel related events, they do not contain all of the fields in a standard
channel snapshot, nor is the description of the fields identical. As such,
the patch opts for documentation for each field, for each event.
Review: https://reviewboard.asterisk.org/r/4350/
ASTERISK-24671 #close
Reported by: Dan Jenkins
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Matthew Jordan [Wed, 21 Jan 2015 13:12:04 +0000 (13:12 +0000)]
apps/app_dial: Don't publish DialEnd twice on unexpected GoSub/Macro values
The Dial application has some interesting options with the mid-call Macro (M)
and GoSub (U) options. If the MACRO_RESULT/GOSUB_RESULT returns specific
values, the Dial application will take some action upon the channels involved
in the dial operation (such as hanging up a particular party, etc.) The Dial
application ensures that a Stasis message is published in the event that
MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial operation, so
that there is a corresponding DialEnd event published in AMI/ARI for the
DialBegin event that preceeded it.
A bug exists where that same DialEnd event will be published on Stasis even if
the value returned in MACRO_RESULT/GOSUB_RESULT is not one that the Dial
application cares about. This causes two DialEnd events to be published - one
with the MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is all
sorts of wrong.
This patch fixes the bug by ensuring that we only publish a DialEnd message to
Stasis if the Dial application's mid-call Macro/GoSub returns something that
Dial cares about.
Review: https://reviewboard.asterisk.org/r/4336
ASTERISK-24682 #close
Reported by: Matt Jordan
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Matthew Jordan [Wed, 21 Jan 2015 13:06:06 +0000 (13:06 +0000)]
main/rtp_engine: Format NTP timestamps as unsigned longs
When the RTCP reports are created, the NTP timestamps are stored as strings,
as JSON does not have an integer type long enough to store the value. However,
on 32-bit systems, a signed long may overflow for some portion of the
timestamp.
This patch corrects the overflow by formatting the timestamps as unsigned
longs.
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Ashley Sanders [Tue, 20 Jan 2015 17:15:54 +0000 (17:15 +0000)]
ARI: Fixed crash that occurred when updating a bridge when the optional query parameter 'name' was not supplied.
Prior to this changeset, posting to the: /ari/bridges/{bridgeId} endpoint without specifying a value for the [name] query parameter, would crash Asterisk if the bridge you are attempting to create (or update) had the same ID as an existing bridge. The internal mechanism of the POST operation interpreted a null value for name, thus resulting in an error condition that crashed Asterisk.
Richard Mudgett [Tue, 20 Jan 2015 16:59:30 +0000 (16:59 +0000)]
CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge.
Calling ast_channel_bridge_peer() cannot be done while holding any channel
locks. The reported issue hit the deadlock in chan_iax2, but an audit of
the ast_channel_bridge_peer() calls found three more locations where the
same deadlock can occur.
* Made CHANNEL(peer), res_fax, and the SNMP agent not call
ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I
had to rework the logic to not hold the channel lock.
* Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done
for legacy reasons that no longer apply.
* Removed the iax.conf forcejitterbuffer option. It is now always enabled
when the jitterbuffer option is enabled. If you put a jitter buffer on a
channel it will be on the channel.
Matthew Jordan [Tue, 20 Jan 2015 02:41:09 +0000 (02:41 +0000)]
contrib/scripts/install_prereq: Don't install 32-bit packages on 64-bit hosts
On Debian based systems, the install_prereq tool uses a search command on
Debian that results in selecting both 64-bit and 32-bit packages. Besides the
waste of disk space, this can actually cause aptitude use 100% of memory on a
VM with 1GB of RAM as it tried to work out all of the 32-bit package
dependencies.
This patch filters out the 32-bit packages on a 64-bit machine, and leaves
32-bit machines alone.
ASTERISK-24048 #close
Reported by: Ben Klang
Tested by: Ben Klang, Matt Jordan
patches:
install_prereq_64-bit_compat.patch uploaded by Ben Klang (License 5876)
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Matthew Jordan [Tue, 20 Jan 2015 02:33:24 +0000 (02:33 +0000)]
app_voicemail: Temp message left after review/hangup with ODBC/IMAP backend
When using ODBC or IMAP storage, temporary files created on the file system
must be disposed of using the DISPOSE macro. The DELETE macro will map to a
deletion function for the backend storage, but does not clean up any local
files created as a result of the operation.
When using voicemail with the operator and review options enabled, pressing
0 to enter the menu, followed by 1 to save the message, followed by any
other DTMF press to delete the message, will result in the temporary file
lingering on the file system.
This patch properly calls DISPOSE after the DELETE. This causes the local
file to be disposed of.
ASTERISK-24288 #close
Reported by: LEI FU
patches:
voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640)
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Mark Michelson [Mon, 19 Jan 2015 18:15:03 +0000 (18:15 +0000)]
Call extension state callbacks at hint creation.
When a hint gets created, any subsequent device or presence
state changes result in extension status events getting sent
out to interested parties. However, at the time of hint creation,
no such event gets sent out, so watchers of extension state are
potentially left in the dark until the first state change after
hint creation.
Patch contributed by John Hardin (License #6512)
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Joshua Colp [Mon, 19 Jan 2015 13:19:11 +0000 (13:19 +0000)]
res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions.
The first thing this patch fixes is UAS dialogs. Previously if a transport was
configured on an endpoint and an inbound session was created there was no guarantee
that requests sent on the dialog would use the correct transport and address
information. This has now been fixed so an explicitly configured transport
is taken into account.
The second thing this patch fixes is res_pjsip_multihomed. The res_pjsip_multihomed
module attempts to determine what transport a message should go out on and what
addressing information should go into the message itself. In a scenario where
multiple transports exist bound to the same IP address but a different port the
code would incorrectly alter the transport and change the message to the wrong
transport. This change makes the res_pjsip_multihomed module smarter so it will
only change the transport and address information in the message when it is
possible and makes sense.
Mark Michelson [Fri, 16 Jan 2015 22:13:23 +0000 (22:13 +0000)]
Fix problem where a hung channel could occur on a failed blind transfer.
Different clients react differently to being told that a blind transfer
has failed. Some will simply send a BYE and be done with it. Others will
attempt to reinvite themselves back onto the call.
In the latter case, we were creating a new channel and then leaving it to
sit forever doing nothing. With this code change, that new channel will
not be created and the dialog with the transferring channel will be cleaned
up properly.
When FAX was developed, apparently the faxregistry.container used to be a
linked list that was converted to an ao2 container. Some of the
replacement ao2 container operations still had explicit lock/unlocks
around them.
Three off nominal code paths in res_fax.c and res_fax_spandsp.c unlock the
channel even though the routine did not lock the channel and other code
paths in the routine do not unlock the channel.
Joshua Colp [Thu, 15 Jan 2015 12:10:22 +0000 (12:10 +0000)]
res_pjsip_outbound_registration: Fix race condition when reloading and listing registrations.
Due to the split of outbound registration state from configuration it is possible during
a reload for a "pjsip show registrations" CLI command to be executed which gets an older
snapshot of the configuration. This configuration may include outbound registrations which
have been removed due to a reload operation occurring at the same time. The code for
printing the outbound registration did not take this into account but now it does.
Matthew Jordan [Thu, 15 Jan 2015 02:19:49 +0000 (02:19 +0000)]
configure: If cross-compiling, assume we have working semaphores
The Asterisk 13 configure.ac checks for HAS_WORKING_SEMAPHORE but does not have
an option for cross-compiling so it fails with an exit. Since we're cross-
compiling, we can't exactly go looking for the header. The semaphore.h header
is relatively common:
* It's part of the POSIX standard
* It's part of GNU C Library
As such, we assume that it will be present when cross-compiling.
As such, this patch defaults "HAS_WORKING_SEMAPHORE" to "1" if cross-compiling
is detected.
If you're cross-compiling to a platform that doesn't support this, then make
sure you re-define this to 0.
Kevin Harwell [Wed, 14 Jan 2015 23:15:23 +0000 (23:15 +0000)]
res_pjsip: make it unloadable
The res_pjsip module was previously unloadable. With this patch it can now
be unloaded.
This patch is based off the original patch on the issue (listed below) by Corey
Farrell with a few modifications. Namely, removed a few changes not required to
make the module unloadable and also fixed a bug that would cause asterisk to
crash on unloading.
This patch is the first step (should hopefully be followed by another/others at
some point) in allowing res_pjsip and the modules that depend on it to be
unloadable. At this time, res_pjsip and some of the modules that depend on
res_pjsip cannot be unloaded without causing problems of some sort.
The goal of this patch is to get res_pjsip and only res_pjsip to be able to
unload successfully and/or shutdown without incident (crashes, leaks, etc...).
Other dependent modules may still cause problems on unload.
Basically made sure, with the patch applied, that res_pjsip (with no other
dependent modules loaded) could be succesfully unloaded and Asterisk could
shutdown without any leaks or crashes that pertained directly to res_pjsip.
Mark Michelson [Wed, 14 Jan 2015 20:39:01 +0000 (20:39 +0000)]
Prevent slow graceful shutdown when outbound publications never started.
The code was missing the case for explicitly destroying an outbound publication
when Asterisk had never actually published anything. The result was that Asterisk
would hang for a while on a graceful shutdown.
With this change, the case is taken into account, and on a graceful shutdown, these
publications are destroyed without the need to actually send a PUBLISH request.
Richard Mudgett [Tue, 13 Jan 2015 18:17:51 +0000 (18:17 +0000)]
app_macro: Don't restore the calling location on a channel redirect.
v11: If a channel redirect to a macro exten of a macro that is active
happens, the redirect location doesn't get executed. Instead the original
macro location is restored and gets reexecuted.
v13: An additional effect happens if a parked call times out to an
extension in the macro that parked the call then the macro is reexecuted
instead of the expected park return location.
* Made not restore the macro calling location on an
AST_SOFTHANGUP_ASYNCGOTO.
* Increased the locked channel range when setting up the macro execution
environment to cover things that should be done while the channel is
locked.
* Removed unnecessary NULL tests before calling ast_free() in
_macro_exec().
Joshua Colp [Tue, 13 Jan 2015 12:09:45 +0000 (12:09 +0000)]
chan_pjsip: Add configure check for 'pjsip_get_dest_info' function.
The 'pjsip_get_dest_info' function is used to determine if the signaling transport
of the dialog is secure or not. This function was added in PJSIP 2.3 and does not
exist in earlier versions.
This configure check allows Asterisk to build and run with older versions at the
loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of
this argument will require upgrading to PJSIP 2.3.
Richard Mudgett [Mon, 12 Jan 2015 19:13:03 +0000 (19:13 +0000)]
AMI: Revert non-backwards compatible changes from earlier commit.
* Reverted the change to astman_send_listack() to not use the listflag
parameter and always set the value to "Start" so the start capitalization
is consistent. Unfortunately changing the case of a returned value is not
a backward compatible change so for now FAXSessions is going to have to
remain inconsistent with all of the other AMI list actions.
* Reverted the minor protocol error fix in action_getconfig() when no
requested categories are found. Each line needs to be formatted as
"Header: text".
Caught by the testsuite.
ASTERISK-24049
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Matthew Jordan [Mon, 12 Jan 2015 18:28:50 +0000 (18:28 +0000)]
configs/samples/features.conf.sample: Document attended transfer DTMF options
The sample config was missing the configuration options for DTMF attended
transfer completion scenarios. The configuration options 'atxferabort',
'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the
appropriate configuration file.
Matthew Jordan [Mon, 12 Jan 2015 18:01:46 +0000 (18:01 +0000)]
main/syslog: Allow dynamic logs, such as security events, to log to the syslog
The security event log uses a dynamic log level (SECURITY) that is registered
with the Asterisk logging core. Unfortunately, the syslog would ignore log
statements that had a dynamic log level associated with them. Because the
syslog cannot handle ad hoc dynamic log levels, this patch treats any dynamic
log entries sent to the syslog as logs with a level of NOTICE.
ASTERISK-20744 #close
Reported by: Michael Keuter
Tested by: Michael L. Young, Jacek Konieczny
patches:
asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by Michael L. Young (license 5026)
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Matthew Jordan [Mon, 12 Jan 2015 15:18:24 +0000 (15:18 +0000)]
funcs/func_curl: Fix memory leak when CURLOPT channel datastore is destroyed
When the channel datastore associated with the usage of CURLOPT on a specific
channel is freed, the underlying structure holding the list of options is not
disposed of. This patch properly frees the structure in the datastore .destroy
callback.
ASTERISK-24672 #close
Reported by: Kristian Hogh
patches:
func_curl-memory-leak.diff uploaded by Kristian Hogh (License 6639)
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When app_bridge grabs a channel and puts it into
a bridge, the channel should then continue where
it left off in the dialplan after the bridge has
ended. Although it stores the current dialplan
location as an after bridge goto on the channel,
it was executing the same priority again instead
of going to the next priority. By swapping the
"specific" version of bridge_set_after_goto with
bridge_set_after_go_on, the next priority in the
dialplan is executed instead.
ASTERISK-24637 #close
Review: https://reviewboard.asterisk.org/r/4322/
Reported by: John Bigelow
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Richard Mudgett [Fri, 9 Jan 2015 18:16:54 +0000 (18:16 +0000)]
AMI: Make AMI actions that generate event lists consistent.
* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
Kinsey Moore [Fri, 9 Jan 2015 14:53:09 +0000 (14:53 +0000)]
res_fax: Add T.38 negotiation timeout option
This change makes the T.38 negotiation timeout configurable via
't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously
hard coded to be 5000 milliseconds.
This change also handles T.38 switch failures by aborting the fax since
in the case where this can happen, both sides have agreed to switch to
T.38 and Asterisk is unable to do so.
George Joseph [Thu, 8 Jan 2015 21:41:02 +0000 (21:41 +0000)]
res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown
If you do a 'core (shutdown|restart) graceful' persistent subscriptions won't
survive. If you do a 'core (shutdown|restart) now' or asterisk terminates for
some reason, they do. Here's why...
When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to
subscribers for each subscription. This not only tells the subscribers that the
dialog/state machine is done, it also frees the last reference to the
subscription tree which causes the persistent subscription to get deleted from
astdb. When asterisk restarts, nothing's left. Just preventing the delete from
astdb doesn't work because we already told the subscriber to terminate the
dialog so we can't restart it even if it was still in astdb. Everything works
OK if asterisk terminates unexpectedly because we never send the 'terminated'
message so on restart, the subscription is still in astdb and the subscriber is
none the wiser.
This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for
persistent connections.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4318/
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Every time a registration started, sip_outbound_registration_response_cb bumps
the ref count on client_state then pushes a handle_registration_response task.
handle_registration_response never unreffed it though. So every time a
registration goes out, the ref count goes up by one.
This patch adds the unreffs to handle_registration_response.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4303/
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George Joseph [Thu, 8 Jan 2015 17:51:36 +0000 (17:51 +0000)]
res_pjsip_outbound_registration: Fix several reload issues
There are 2 issues with reloading registrations...
1. The 'can_reuse_registration' test wasn't considering the intervals or
expiration in its determination of whether a registration changed or not so if
you changed any of the intervals or the expiration and reloaded, the object
would get reloaded but the actual timers wouldn't change.
can_reuse_registration now does a sorcery diff on the old and new objects
instead of discretely testing certain fields. Now if you change expiration for
instance, and reload, the timer is updated and re-registration will occur on the
new value.
2. If you mung up your password on an outbound registration you get a permanent
failure. If you fix the password (on the outbound_auth object) and reload,
nothing tells outbound_registration to try again because the registration itself
didn't change. This patch adds an observer on the "auth" object type and if any
auth changes, existing registration states are searched and those in a
REJECTED_PERMANENT state are retried.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4304/
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Kinsey Moore [Wed, 7 Jan 2015 21:26:48 +0000 (21:26 +0000)]
ARI: Allow usage of ASYNCGOTO with Stasis()
When the AMI Redirect action is used with a channel bridged inside
Stasis() and not running a pbx, the channel is hung up instead of
proceeding to the desired location in dialplan. This change allows
such channels to be Redirected properly by detecting the operation
used by Redirect (ASYNCGOTO) and using the code already established
for functionality of the ARI channel continue operation.
George Joseph [Wed, 7 Jan 2015 18:17:42 +0000 (18:17 +0000)]
res_pjsip_exten_state: Change 'does not exist' warning to notice
The 'new_subscribe: Extension <> does not exist or has no associated hint'
is a config issue and doesn't need to clutter up logs with warnings.
Changed to notice.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4307/
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George Joseph [Wed, 7 Jan 2015 18:15:02 +0000 (18:15 +0000)]
res_pjsip_mwi: Change "MWI Subscription failed" message from warning to notice
The "MWI Subscription failed" message means the client is trying to subscribe
to a mailbox that doesn't exist. There's no need to clutter up logs with
warnings for a client misconfiguration so I changed it to a notice.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4306/
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George Joseph [Wed, 7 Jan 2015 17:54:13 +0000 (17:54 +0000)]
func_config: Add ability to retrieve specific occurrence of a variable
I guess nobody uses templates with AST_CONFIG because today if you have a
context that inherits from a template and you call AST_CONFIG on the context,
you'll get the value from the template even if you've overridden it in the
context. This is because AST_CONFIG only gets the first occurrence which is
always from the template.
This patch adds an optional 'index' parameter to AST_CONFIG which lets you
specify the exact occurrence to retrieve, or '-1' to retrieve the last.
The default behavior is the current behavior.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4313/
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Mark Michelson [Wed, 7 Jan 2015 17:45:56 +0000 (17:45 +0000)]
Fix ability to perform a remote attended transfer with PJSIP.
This fix has two parts:
* Corrected an error message to properly state that external_replaces is an extension. The
error message also prints what dialplan context the external_replaces extension was being
looked for in.
* Corrected the printing of the Replaces: header in an INVITE request. We were duplicating
"Replaces: " in the header.
George Joseph [Wed, 7 Jan 2015 16:56:59 +0000 (16:56 +0000)]
config: Add option to NOT preserve effective context when changing a template
Let's say you have a template T with variable VAR1 = ON and you have a
context C(T) that doesn't specify VAR1. If you read C, the effective value
of VAR1 is ON. Now you change T VAR1 to OFF and call
ast_config_text_file_save. The current behavior is that the file gets
re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this
is a bug. It's preserving the effective state of C even though I didn't
specify C/VAR1 in th first place. I believe the behavior should be that if
I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should
continue to follow the inherited state. Now, if I DID explicitly specify
C/VAR1, the it should be preserved even if the template changes.
Even though I think the existing behavior is a bug, it's been that way forever
so I'm not changing it. Instead, I've created ast_config_text_file_save2()
that takes a bitmask of flags, one of which is to preserve the effective context
(the current behavior). The original ast_config_text_file_save calls *2 with
the preserve flag. If you want the new behavior, call *2 directly without a
flag.
I've also updated Manager UpdateConfig with a new parameter
'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior
with UpdateConfig, set 'PreserveEffectiveContext: no'.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4297/
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Matthew Jordan [Tue, 6 Jan 2015 22:46:43 +0000 (22:46 +0000)]
Blocked revisions 430252
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contrib/ast-db-manage: Correct down_revision path for user_eq_phone
When the user_eq_phone patch was backported to 13, it referenced the downward
revision that the PJSIP optimistic encryption option also references. This
creates a multi-path upgrade Exception when generating the SQL files.
This patch corrects this in the 13 branch. Note that trunk, which already
contained both of these features, is unaffected by this problem.
George Joseph [Tue, 6 Jan 2015 17:53:42 +0000 (17:53 +0000)]
res_pjsip_mwi: Change warning to notice
When res_pjsip loads and an endpoint auto-subscribes a mailbox for mwi,
if a contact hasn't registered yet, res_pjsip_mwi spits out a warning.
This is a perfectly normal situation though and doesn't require something
as serious as a warning. It's also self correcting. The device will start
getting mwi as soon as it registers.
This patch changes the warning to a notice.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4314/
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George Joseph [Tue, 6 Jan 2015 17:43:16 +0000 (17:43 +0000)]
outbound_registration: Add 'pjsip send register' and update 'send unregister'
The current behavior of 'pjsip send unregister' is to send the unregister
(REGISTER with 0 exp) but let the next scheduled register proceed normally.
I don't think that's a good idea. If you unregister, it should stay
unregistered until you decide to start registrations again. So this patch
just adds a cancel_registration call to the current unregister_task to
cancel the timer.
Of course, now you need a way to start registration again so I've added
a 'pjsip send register' command that unregisters and cancels any existing
registration (the same as send unregister), then sends an immediate
registration and starts the timer back up again.
Both changes also ripple to AMI. There's a new PJSIPRegister command.
There's no harm in calling either command repeatedly. They don't care
about the actual state.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4301/
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George Joseph [Tue, 6 Jan 2015 17:29:33 +0000 (17:29 +0000)]
pjsip cli: Fix sorting of contacts for 'pjsip list contacts'
For some reason I was using a hash container instead of a list to gather the
contacts for 'pjsip list/show contacts' so even though I had a sort function,
the output wasn't sorted. This patch just changes the hash container to a
list container and the contacts now appear sorted in the CLI.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4305/
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bridge: avoid leaking channel during blond transfer pt2
A blond transfer to a failed destination, when followed
by a recall attempt, lead to a leak of the reference to
the destination channel. In addition to correcting the
regression on the previous attempt (r429826) this fixes
the leak and two additional reference leaks on failures
of bridge_import.
Joshua Colp [Mon, 5 Jan 2015 17:53:42 +0000 (17:53 +0000)]
pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions.
The PJSIP_AOR dialplan function allows inspection of configured AORs including
what contacts are currently bound to them.
The PJSIP_CONTACT dialplan function allows inspection of contacts in existence.
These can include both externally added (by way of registration) or permanent
ones.
In r428708 additional codecs were added including
a payload type of 128 which is outside of nominal
range of 0-127. This change moves changes 128 to
96 to avoid causing a pjsip assertion when making
a call to an endpoint configured with allow=all.
Kinsey Moore [Mon, 29 Dec 2014 13:14:19 +0000 (13:14 +0000)]
PJSIP: Update transport method documentation
This updates the documentation for the 'method' configuration option to
be more verbose about the behaviors of values 'unspecified' and
'default'. They do exactly the same thing which is to select the
default as defined by PJSIP which is currently TLSv1.
George Joseph [Tue, 23 Dec 2014 23:19:30 +0000 (23:19 +0000)]
pjsip_options: Fix continued qualifies after endpoint/aor deletion
If you remove an endpoint/aor from pjsip.conf then do a core reload,
qualifies will continue even though the object are gone. This happens
because nothing clears out the qualify tasks.
This patch unschedules all existing qualify tasks before scheduling
new ones on reload.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4290/
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Richard Mudgett [Mon, 22 Dec 2014 21:20:11 +0000 (21:20 +0000)]
DTMF atxfer: Setup recall channels as if the transferee initiated the call.
After the initial DTMF atxfer call attempt to the transfer target fails to
answer during a blonde transfer, the recall callback channels do not get
setup with information from the initial transferrer channel. As a result,
the recall callback to the transferrer does not have callid, channel
variables, datastores, accountcode, peeraccount, COLP, and CLID setup. A
similar situation happens with the recall callback to the transfer target
but it is less visible. The recall callback to the transfer target does
not have callid, channel variables, datastores, accountcode, peeraccount,
and COLP setup.
* Added missing information to the recall callback channels before
initiating the call. callid, channel variables, datastores, accountcode,
peeraccount, COLP, and CLID
* Set callid of the transferrer channel on the DTMF atxfer controller
thread attended_transfer_monitor_thread().
* Added missing channel unlocks and props unref to off nominal paths in
attended_transfer_properties_alloc().
ASTERISK-23841 #close
Reported by: Richard Mudgett
Richard Mudgett [Mon, 22 Dec 2014 20:08:35 +0000 (20:08 +0000)]
queue_log: Post QUEUESTART entry when Asterisk fully boots.
The QUEUESTART log entry has historically acted like a fully booted event
for the queue_log file. When the QUEUESTART entry was posted to the log
was broken by the change made by ASTERISK-15863.
* Made post the QUEUESTART queue_log entry when Asterisk fully boots.
This restores the intent of that log entry and happens after realtime has
had a chance to load.
Matthew Jordan [Mon, 22 Dec 2014 15:40:27 +0000 (15:40 +0000)]
chan_sip: Send CANCEL via original INVITE destination even after UPDATE request
Given the following scenario:
* Three SIP phones (A, B, C), all communicating via a proxy with Asterisk
* A call is established between A and B. B performs a SIP attended transfer of
A to C. B sets the call on hold (A is hearing MOH) and dials the extension of
C. While phone C is ringing, B transfers the call (that is, what we typically
call a 'blond transfer').
* When the transfer completes, A hears the ringing of phone C, while B is idle.
In the SIP messaging for the above scenario, a REFER request is sent to
transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an
UPDATE request to phone C to update party information. This update is sent
directly to phone C, not through the intervening proxy. This has the unfortunate
side effect of providing route information, which is then set on the sip_pvt
structure for C. If someone (e.g. B) is trying to get the call back (through a
directed pickup), Asterisk will send a CANCEL request to C. However, since we
have now updated the route set, the CANCEL request will be sent directly to C
and not through the proxy. The phone ignores this CANCEL according to RFC3261
(Section 9.1).
This patch updates reqprep such that the route is not updated if an UPDATE
request is being sent while the INVITE state is INV_PROCEEDING or
INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent
to the correct location.
Matthew Jordan [Mon, 22 Dec 2014 14:33:24 +0000 (14:33 +0000)]
presencestate: Allow channel drivers to provide presence state information
This patch adds the ability for channel drivers to supply presence information
in a similar manner to device state. The patch does not provide any channel
driver implementations, but it does provide the core infrastructure necessary
for channel drivers to provide such information.
The core handles multiple providers of presence state information. Ordering
of presence state is as follows:
INVALID < NOT_SET < AVAILABLE < UNAVAILABLE < CHAT < AWAY < XA < DND
Each provider can trump the previous if it provides a presence state that
supercedes a previous one.
Matthew Jordan [Mon, 22 Dec 2014 02:35:05 +0000 (02:35 +0000)]
app_confbridge: Add the ability to pass options/command to MixMonitor
This patch adds the ability to pass options and a command to MixMontor when
recording a conference using ConfBridge.
New options are -
* record_options: Options to MixMontor, eg: m(), W() etc.
* record_command: The command to execute when recording is over.
* record_file_timestamp: Append the start time to the file name.
These options can also be used with the CONFBRIDGE function, e.g.,
Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME}))
George Joseph [Mon, 22 Dec 2014 00:17:49 +0000 (00:17 +0000)]
res_pjsip_phoneprovi_provider: Fix reload
Reloading wasn't working correctly because on a reload, the sorcery apply
handler was never being called for unchanged users. So, instead of using
an apply handler, I'm now iterating over all users. Works much more reliably.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4288/
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Joshua Colp [Sat, 20 Dec 2014 20:57:47 +0000 (20:57 +0000)]
acl: Fix reloading of configuration if configuration file does not exist at startup.
The named ACL code incorrectly destroyed the config options information if loading
of the configuration file failed at startup. This would result in reloading
also failing even if a valid configuration file was put in place.
ASTERISK-23733 #close
Reported by: Richard Kenner
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Richard Mudgett [Fri, 19 Dec 2014 17:34:33 +0000 (17:34 +0000)]
chan_dahdi: Don't ignore setvar when using configuration section scheme.
When the configuration section scheme of chan_dahdi.conf is used (keyword
dahdichan instead of channel) all setvar= options are completely ignored.
No variable defined this way appears in the created DAHDI channels.
* Move the clearing of setvar values to after the deferred processing of
dahdichan.
bridge: avoid leaking channel during blond transfer
After a blond transfer (start attended and hang up)
to a destination that also hangs up without answer,
the Local;1 channel was leaked and would show up on
core show channels. This was happening because the
attended state blond_nonfinal_enter() resetting the
props->transfer_target to null while releasing it's
own reference, which would later prevent props from
releasing another reference during destruction. The
change made here is simply to not assign the target
to NULL.
ASTERISK-24513 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4262/
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Kevin Harwell [Thu, 18 Dec 2014 15:55:03 +0000 (15:55 +0000)]
res_pjsip_sdp_rtp: wrong bridge chosen when the DTMF mode is not compatible
A native rtp bridge was being chosen (it shouldn't have been) when using two
pjsip channels with incompatible DTMF modes. This patch sets the rtp instance
property, AST_RTP_PROPERTY_DTMF, for the appropriate DTMF mode(s) for pjsip.
It was not being set before, meaning all DTMF modes for pjsip were being treated
as compatible, thus native bridging would be chosen as the bridge type when it
shouldn't have been.
Mark Michelson [Thu, 18 Dec 2014 15:40:13 +0000 (15:40 +0000)]
Prevent potential infinite outbound authentication loops in registration.
Prior to this patch, Asterisk would always respond to 401 responses to
registration attempts by trying to provide a registration with authentication
credentials. Even if subsequent attempts were rejected with 401 responses,
Asterisk would continue this behavior. If authentication credentials were
incorrect, this could continue forever.
With this patch, we keep track of whether we have attempted authentication
on an outbound registration attempt. If we already have, we don not try
again until the next attempt. This prevents the infinite loop scenario.
Mark Michelson [Thu, 18 Dec 2014 15:18:45 +0000 (15:18 +0000)]
Prevent possible race condition on dual redirect of channels in the same bridge.
The AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent bridges from
prematurely acting on orphaned channels in bridges. The problem with the AMI
redirect action was that it was setting this flag on channels based on the presence
of a PBX, not whether the channel was in a bridge. Whether a channel has a PBX
is irrelevant, so the condition has been altered to check if the channel is in a
bridge.
Mark Michelson [Thu, 18 Dec 2014 14:50:06 +0000 (14:50 +0000)]
Ensure the correct value is returned for CHANNEL(pjsip, secure)
Prior to this patch, we were using the PJSIP dialog's secure flag
to determine if a secure transport was being used. Unfortunately,
the dialog's secure flag was only set if a SIPS URI were in use,
as required by RFC 3261 sections 12.1.1 and 12.1.2. What we're interested
in is not dialog security, but transport security. This code change
switches to a model where we use the dialog's target URI to determine
what transport would be used to communicate, and then check if that
transport is secure.
George Joseph [Thu, 18 Dec 2014 00:11:24 +0000 (00:11 +0000)]
res_pjsip_config_wizard: fix unload SEGV
If certain pjsip modules aren't loaded, the wizard causes a SEGV
when it unloads. Added a check for the presense of the object
type wizard before trying to clean it up.
Tested-by: George Joseph
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That works for low ascii characters, but for the high range that yields
e.g. FFFFFFC3 when C3 is expected.
This changeset:
- fixes those casts to use the 'hh' unsigned char modifier instead
- consistently uses %02x instead of %2.2x (or other non-standard usage)
- adds a few 'h' modifiers in various places
- fixes a 'replcaes' typo
- dev/urandon typo (in 13+ patch)
Review: https://reviewboard.asterisk.org/r/4263/
ASTERISK-24619 #close
Reported by: Stefan27 (on IRC)
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Joshua Colp [Tue, 16 Dec 2014 16:39:47 +0000 (16:39 +0000)]
chan_sip: Allow T.38 switch-over when SRTP is in use.
Previously when SRTP was enabled on a channel it was not possible
to switch to T.38 as no crypto attributes would be present.
This change makes it so it is now possible. If a T.38 re-invite
comes in SRTP is terminated since in practice you can't encrypt
a UDPTL stream. Now... if we were doing T.38 over RTP (which
does exist) then we'd have a chance but almost nobody does that so
here we are.
ASTERISK-24449 #close
Reported by: Andreas Steinmetz
patches:
udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523)
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Joshua Colp [Tue, 16 Dec 2014 15:44:43 +0000 (15:44 +0000)]
res_pjsip_t38: Fix T.38 failure when peer reinvites immediately.
If a remote endpoint reinvites to T.38 immediately the state machine
will go into a peer reinvite state. If a T.38 capable application
(such as ReceiveFax) queries it will receive this state. Normally
the application will then indicate so that the channel driver will
queue up the T.38 offer previously received. Once it receives this
offer the application will act normally and negotiate.
The res_pjsip_t38 module incorrectly partially squashed this indication.
This would cause the application to think the request had failed when
in reality it had actually worked.
This change makes it so that no T.38 control frames (or indications)
are squashed.
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George Joseph [Mon, 15 Dec 2014 17:08:24 +0000 (17:08 +0000)]
res_pjsip_config_wizard: Allow streamlined config of common pjsip scenarios
res_pjsip_config_wizard
------------------
* This is a new module that adds streamlined configuration capability for
chan_pjsip. It's targetted at users who have lots of basic configuration
scenarios like 'phone' or 'agent' or 'trunk'. Additional information
can be found in the sample configuration file at
config/samples/pjsip_wizard.conf.sample.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4190/
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Mark Michelson [Mon, 15 Dec 2014 15:48:47 +0000 (15:48 +0000)]
Activate persistent subscriptions when they are recreated.
Prior to this change, recreating persistent subscriptions would
create the subscription but would not activate it. This led to subscriptions
being listed in the "NULL" state by diagnostics and not sending NOTIFYs
when expected.
Richard Mudgett [Fri, 12 Dec 2014 23:49:36 +0000 (23:49 +0000)]
DEBUG_THREADS: Fix regression and lock tracking initialization problems.
This patch started with David Lee's patch at
https://reviewboard.asterisk.org/r/2826/ and includes a regression fix
introduced by the ASTERISK-22455 patch.
The initialization of a mutex's lock tracking structure was not protected
in a critical section. This is fine for any mutex that is explicitly
initialized, but a static mutex may have its lock tracking double
initialized if multiple threads attempt the first lock simultaneously.
* Added a global mutex to properly serialize initialization of the lock
tracking structure. The painful global lock can be mitigated by adding a
double checked lock flag as discussed on the original review request.
* Defer lock tracking initialization until first use.
* Don't be "helpful" and initialize an uninitialized lock when
DEBUG_THREADS is enabled. Debug code is not supposed to fix or change
normal code behavior. We don't need a lock initialization race that would
force a re-setup of lock tracking. Lock tracking already handles
initialization on first use.
* Properly handle allocation failures of the lock tracking structure.
* No need to initialize tracking data in __ast_pthread_mutex_destroy()
just to turn around and destroy it.
The regression introduced by ASTERISK-22455 is the result of manipulating
a pthread_mutex_t struct outside of the pthread library code. The
pthread_mutex_t struct seems to have a global linked list pointer member
that can get changed by other threads. Therefore, saving and restoring
the contents of a pthread_mutex_t struct is a bad thing.
Thanks to Thomas Airmont for finding this obscure regression.
* Don't overwrite the struct ast_lock_track.reentr_mutex member to restore
tracking data in __ast_cond_wait() and __ast_cond_timedwait(). The
pthread_mutex_t struct must be treated as a read-only opaque variable.
Miscellaneous other items fixed by this patch:
* Match ast_suspend_lock_info() with ast_restore_lock_info() in
__ast_cond_timedwait().
* Made some uninitialized lock sanity checks return EINVAL and try a
DO_THREAD_CRASH.
Matthew Jordan [Fri, 12 Dec 2014 22:54:02 +0000 (22:54 +0000)]
res/res_agi: Make Verbose message for 'stream file' match other playbacks
The Verbose message displayed when a file is played back via 'stream file'
was formatted differently than other playbacks:
* It didn't include the channel name
* It didn't include the channel language
It does, however, include the playback offset as well as any escape digits.
That information was kept; however, this patch updates the formatting to more
closely match the Verbose messages displayed when a file is played back by
'control stream file', Playback, ControlPlayback, or any other file playback
operation.
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Joshua Colp [Fri, 12 Dec 2014 17:01:42 +0000 (17:01 +0000)]
media: Fix crash when determining sample count of a frame during shutdown.
When shutting down Asterisk the codecs are cleaned up. As a result anything
attempting to get a codec based on ID or details will find that no codec
exists. This currently occurs when determining the sample count of a frame.
This code did not take this situation into account.
This change fixes this by getting the codec directly from the format and
eliminates the lookup. This is both faster and also provides a guarantee
that the codec will exist and will be valid.
Kevin Harwell [Fri, 12 Dec 2014 15:31:38 +0000 (15:31 +0000)]
chan_pjsip: Race between channel answer and bridge setup when using direct media
When direct media is enabled and a pjsip channel is answered a race would occur
between the handling of the answer and bridge setup. Sometimes the media
negotiation would take place after the native bridge was setup. This resulted
in a NULL media address, which in turn resulted in Asterisk using its address
as the remote media address when sending a reinvite. This patch makes the
chan_pjsip answer handler synchronous thus alleviating the race condition (the
bridge won't start setting things up until after it returns).
ASTERISK-24563 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4257/
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David M. Lee [Fri, 12 Dec 2014 15:03:16 +0000 (15:03 +0000)]
Fix crash for sorcery misconfigs
res_pjsip_outbound_publish was missing the CHECK_PJSIP_MODULE_LOADED()
call in load_module, and would crash with a segfault if res_pjsip
declined to load.
Kinsey Moore [Fri, 12 Dec 2014 14:12:38 +0000 (14:12 +0000)]
PJSIP: Allow use of 'inactive' streams for hold
This allows use of the 'inactive' stream direction identifier to be
used for hold where 'sendonly' is normally used. Some Seimens phones
use 'inactive' and this change allows music on hold to operate
properly.
Review: https://reviewboard.asterisk.org/r/4252/
Reported by: Steve Pitts
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projects / thirdparty / asterisk.git / log
