Corey Farrell [Tue, 14 Oct 2014 16:47:02 +0000 (16:47 +0000)]
res_fax: Fix reference leak caused by gateway sessions
Fax gateway session objects can be re-used, causing the
same gateway session to be added to faxregistry.container
more than once. This change causes fax_session_new to
remove the reserved session from the container before
it's id is changed, ensuring it's possible for the
session to be freed.
Richard Mudgett [Tue, 14 Oct 2014 16:43:33 +0000 (16:43 +0000)]
stasis_channels.c: Resolve unfinished Dials when doing masquerades (Part 2)
Masquerades into and out of channels that are involved in a dial operation
don't create the expected dial end event. The missing dial end event goes
against the model for things like CDRs and generating Dial end manager
actions and such.
There are four cases:
1) A channel masquerades into the caller channel. The case happens when
performing a blonde transfer using the channel driver's protocol.
2) A channel masquerades into a callee channel. The case happens when
performing a directed call pickup.
3) The caller channel masquerades out of dial. The case happens when
using the Bridge application on the caller channel.
4) A callee channel masquerades out of dial. The case happens when using
the Bridge application on a peer channel.
As it turned out, all four cases need to be handled instead of just the
first one.
ASTERISK-24237
Reported by: Richard Mudgett
ASTERISK-24394 #close
Reported by: Richard Mudgett
Corey Farrell [Tue, 14 Oct 2014 16:20:59 +0000 (16:20 +0000)]
res_fax: Resolve module reference leak caused by reserved sessions
Remove reference to module providing reserved session after
adding a reference to the final module. This re-reference
is done to ensure that module references are correct even
if the final session selects a different module than the
reserved session.
George Joseph [Mon, 13 Oct 2014 16:12:17 +0000 (16:12 +0000)]
manager/config: Support templates and non-unique category names via AMI
This patch provides the capability to manipulate templates and categories
with non-unique names via AMI.
Summary of changes:
GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list
of name_regex=value_regex expressions which will cause only categories whose
variables match all expressions to be considered. The special variable name
TEMPLATES can be used to control whether templates are included. Passing
'include' as the value will include templates along with normal categories.
Passing 'restrict' as the value will restrict the operation to ONLY templates.
Not specifying a TEMPLATES expression results in the current default behavior
which is to not include templates.
UpdateConfig: NewCat now includes options for allowing duplicate category
names, indicating if the category should be created as a template, and
specifying templates the category should inherit from. The rest of the
actions now accept a filter string as defined above. If there are non-unique
category names, you can now update specific ones based on variable values.
To facilitate the new capabilities in manager, corresponding changes had to be
made to config, most notably the addition of filter criteria to many of the
APIs. In some cases it was easy to change the references to use the new
prototype but others would have required touching too many files for this
patch so a wrapper with the original prototype was created. Macros couldn't
be used in this case because it would break binary compatibility with modules
such as res_digium_phone that are linked to real symbols.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4033/
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Walter Doekes [Sun, 12 Oct 2014 08:17:08 +0000 (08:17 +0000)]
chan_sip: Fix so asterisk won't send reINVITE after a BYE.
After a reINVITE glare situation, Asterisk would re-send the reINVITE
even though the call had been hung up in the mean time. This patch
unschedules the reinvite when handling the BYE.
ASTERISK-22791 #close
Reported by: Paolo Compagnini
Tested by: Paolo Compagnini
Review: https://reviewboard.asterisk.org/r/4056/
(testcase is in review r4055)
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Walter Doekes [Sun, 12 Oct 2014 07:57:06 +0000 (07:57 +0000)]
build: Relax badshell tilde test to allow for ~ in middle of DESTDIR.
The main Makefile has a target test called 'badshell' that tests if
DESTDIR does not happen to have an an-expanded tilde (~). This might
be the case if you run: make install DESTDIR=~/somewhere/
That test also disallowed valid tildes in directory names. The test is
now changed to only trigger on a tilde at the start of the path.
George Joseph [Sat, 11 Oct 2014 21:09:53 +0000 (21:09 +0000)]
res_phoneprov: Cleanup module load error handling
Tested module load/reload interaction between res_phoneprov and
res_pjsip_phoneprov_provider in cases where res_phoneprov didn't
load correctly (usually misconfiguration or missing phoneprov.conf)
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4069/
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Joshua Colp [Fri, 10 Oct 2014 20:48:46 +0000 (20:48 +0000)]
bridge: During a smart bridge operation provide a more complete bridge to the old technology.
When a smart bridge operation occurs and a bridge transitions from one
technology to another the old technology is provided the channels formerly
in it and told that they are leaving. Unfortunately the bridge provided
along with them is incomplete. The bridge, despite there being channels in it,
contains none. This forces technology implementations to have additional
logic when channels are leaving or to store their own duplicated
state.
This change makes the bridge more complete so it contains the expected
channels. Now that the bridge is complete special logic within
bridge_native_rtp is no longer needed and has been removed.
Matthew Jordan [Fri, 10 Oct 2014 14:31:42 +0000 (14:31 +0000)]
res/res_phoneprov: Bail on registration if res_phoneprov didn't load
If res_phoneprov failed to fully load (due to not being configured), the
providers container will be NULL. If a module attempts to register a phone
provisioning provider, it should check for the presence of the container.
If there is no providers container, it should return an error.
This patch makes the ast_phoneprov_provider_register function do that...
otherwise this would be a silly commit message.
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Kinsey Moore [Fri, 10 Oct 2014 13:03:18 +0000 (13:03 +0000)]
CallerID: Fix parsing regression
This fixes a regression in callerid parsing introduced when another bug
was fixed. This bug occurred when the name was composed entirely of
DTMF keys and quoted without a number section (<>).
ASTERISK-24406 #close
Reported by: Etienne Lessard
Tested by: Etienne Lessard
Patches:
callerid_fix.diff uploaded by Kinsey Moore
Review: https://reviewboard.asterisk.org/r/4067/
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Joshua Colp [Fri, 10 Oct 2014 12:10:53 +0000 (12:10 +0000)]
res_pjsip_nat: Place source port into rport of responses if 'force_rport' is on.
When the 'force_rport' option is enabled the behavior should be the same
as if the remote side placed rport into the message themselves. Therefore
any responses we send should include the source port of the request in the
rport of the Via header.
#SIPit31
ASTERISK-24387 #close
Reported by: Matt Jordan
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Walter Doekes [Fri, 10 Oct 2014 07:34:50 +0000 (07:34 +0000)]
chan_sip: Fix dialog leak resulting from missing ACK to re-INVITE.
If a device re-INVITEs at the same time as the dialog is hung up, and
if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would
fail to destroy the dialog after a while. This resulted in (most
prominently) file handle leaks.
Kevin Harwell [Thu, 9 Oct 2014 21:39:12 +0000 (21:39 +0000)]
res_rtp_asterisk: Crash if no candidates received for component
When starting ice if there is not at least one remote ice candidate with an RTP
component asterisk will crash. This is due to an assertion in pjnath as it
expects at least one candidate with an RTP component. Added a check to make
sure at least one candidate contains an RTP component and at least one candidate
has an RTCP component.
George Joseph [Thu, 9 Oct 2014 17:46:23 +0000 (17:46 +0000)]
res_phoneprov: Refactor phoneprov to allow pluggable config providers
This patch makes res_phoneprov more modular so other modules (like pjsip)
can provide configuration information instead of res_phoneprov relying solely
on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API
is now exposed which allows config providers to register themselves, set
defaults (server profile, etc) and add user extensions.
* ast_phoneprov_provider_register registers the provider and provides callbacks
for loading default settings and loading users.
* ast_phoneprov_provider_unregister clears the defaults and users.
* ast_phoneprov_add_extension should be called once for each user/extension
by the provider's load_users callback to add them.
* ast_phoneprov_delete_extension deletes one extension.
* ast_phoneprov_delete_extensions deletes all extensions for the provider.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/3970/
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Walter Doekes [Thu, 9 Oct 2014 08:10:35 +0000 (08:10 +0000)]
safe_asterisk: Don't automatically exceed MAXFILES value of 2^20.
On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided
by two can exceed the per-process file limit of 2^20. This patch
ensures the value is capped.
(Patch cleaned up by me.)
ASTERISK-24011 #close
Reported by: Michael Myles
Patches:
safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626)
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Joshua Colp [Wed, 8 Oct 2014 18:47:32 +0000 (18:47 +0000)]
res_rtp_asterisk: Allow only UDP ICE candidates.
The underlying library, pjnath, that res_rtp_asterisk uses for ICE
support does not have support for ICE-TCP. As candidates are
passed through directly to it this can cause error messages to occur
when it receives something unexpected (such as a TCP candidate).
This change merely ignores all non-UDP candidates so they never
reach pjnath.
Kinsey Moore [Wed, 8 Oct 2014 14:54:54 +0000 (14:54 +0000)]
Indexer: Format message types may not exist
In Asterisk 13+, any given message type is not guaranteed to exist even
if Asterisk comes up correctly since creation of the message type could
be declined. The indexer should not prevent Asterisk from starting
under these conditions.
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Kinsey Moore [Tue, 7 Oct 2014 20:33:29 +0000 (20:33 +0000)]
Stasis: Only log errors for non-declined types
When message type creation is declined via stasis.conf, certain
operations log errors assuming that the declined type is being used
before initialization or after destruction. These error messages get
quite spammy for oft used message types and should not be logged in the
first place since the message type is validly NULL.
Reported by: Matt DiMeo
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Prior to this patch, the auth_reject_permanent parameter was not initialized on
the registration client state, leading to the parameter being disabled
regardless of the value specified in pjsip.conf.
This patch initialized the setting on the registration client state to the
provided configuration value.
ASTERISK-24398 #close
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Matthew Jordan [Mon, 6 Oct 2014 18:39:54 +0000 (18:39 +0000)]
message: Don't close an AMI connection on SendMessage action error
If SendMessage encounters an error (such as incorrect input provided to the
action), it will currently return -1. Actions should only return -1 if the
connection to the AMI client should be closed. In this case, SendMessage
causing the client to disconnect is inappropriate.
This patch causes the action to return 0, which simply causes the action to
fail.
Review: https://reviewboard.asterisk.org/r/4024
ASTERISK-24354 #close
Reported by: Peter Katzmann
patches:
sendMessage.patch uploaded by Peter Katzmann (License 5968)
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Richard Mudgett [Mon, 6 Oct 2014 15:41:32 +0000 (15:41 +0000)]
features.c: Fix lingering channel ref while Bridge() application is active.
Using the Bridge application to bridge a channel that is executing an
applicaiton such as Wait results in a lingering Surrogate channel in the
CLI "core show channels" output even though it has already hungup.
* Fix bridge_exec() to not hold onto the current_dest_chan ref once it has
been put into the bridge.
Matthew Jordan [Mon, 6 Oct 2014 01:01:43 +0000 (01:01 +0000)]
res_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpoint
An OPTIONS request that is sent to Asterisk but not to a specific endpoint is
currently sent a 404 in response. This is because, not surprisingly, an empty
extension is never going to be found in the dialplan.
This patch makes it so that we only attempt to look up the endpoint in the
dialplan if it is specified in the OPTIONS request URI.
#SIPit31
ASTERISK-24370 #close
Reported by: Matt Jordan
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Matthew Jordan [Mon, 6 Oct 2014 00:53:37 +0000 (00:53 +0000)]
pjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels
Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health.
It will treat the channels as a PJSIP channel, eventually hitting an ao2 error,
FRACKing on assertion error, and quite likely crashing.
This patch adds checks to the read/write callbacks that ensure that the channel
technology is of type 'PJSIP' before attempting to operate on the channel.
#SIPit31
ASTERISK-24382 #close
Reported by: Matt Jordan
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Matthew Jordan [Mon, 6 Oct 2014 00:31:48 +0000 (00:31 +0000)]
res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message
When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable
transport, it is possible (although it shouldn't occur) for pjproject to pass
up an rdata object with a NULL msg in the msg_info. Needless to say, things
that attempt to dereference this are in for a rough ride.
In particular, this caused crashes in three different locations, all of which
are 'low level' enough to intercept an rdata object early in processing:
Matthew Jordan [Mon, 6 Oct 2014 00:13:58 +0000 (00:13 +0000)]
res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions
A subscription that has been persisted can - for various reasons - fail to be
re-created on startup. This patch resolves a number of crashes that occurred
when a subscription cannot be re-created on several off-nominal paths.
#SIPit31
ASTERISK-24368 #close
Reported by: Matt Jordan
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Richard Mudgett [Fri, 3 Oct 2014 19:42:54 +0000 (19:42 +0000)]
audiohooks: Reevaluate the bridge technology when an audiohook is added or removed.
Adding a mixmonitor to a channel causes the bridge to change technologies
from native to simple_bridge so the call can be recorded. However, when
the mixmonitor is stopped the bridge does not switch back to the native
technology.
* Added unbridge requests to reevaluate the bridge when a channel
audiohook is removed.
* Moved the unbridge request into ast_audiohook_attach() ensure that the
bridge reevaluates whenever an audiohook is attached. This simplified the
mixmonitor and chan_spy start code as well.
* Added defensive code to stop_mixmonitor_full() in case additional
arguments are ever added to the StopMixMonitor application.
* Made ast_framehook_detach() not do an unbridge request if the framehook
does not exist.
* Made ast_framehook_list_fixup() do an unbridge request if there are any
framehooks. Also simplified the loop.
Richard Mudgett [Fri, 3 Oct 2014 18:54:53 +0000 (18:54 +0000)]
app_queue: Add dialplan function to get the channel name at the specified position in a queue.
The QUEUE_GET_CHANNEL function returns the caller's channel name at the
specified position in a queue.
QUEUE_GET_CHANNEL(<queuename>[,<position>])
The queue position parameter defaults to 1 if not specified.
Noop(${QUEUE_GET_CHANNEL(queuename, 2)})
"SIP/peer-00000002", if queue exist and have at least 2 callers
Noop(${QUEUE_GET_CHANNEL(queuename, 1)})
Noop(${QUEUE_GET_CHANNEL(queuename)})
"SIP/peer-00000000", if queue exist and have at least 1 caller
ASTERISK-24365 #close
Reported by: Kristian Hogh
Patches:
queue_get_firstchannel.patch (license #6639) patch uploaded by Kristian Hogh
rb4035.patch (license #6639) patch uploaded by Kristian Hogh
Patch morphed from QUEUE_GET_FIRSTCHANEL to the more general QUEUE_GET_CHANNEL
on reviewbord.
Richard Mudgett [Fri, 3 Oct 2014 17:47:42 +0000 (17:47 +0000)]
chan_pjsip: Fix deadlock when masquerading PJSIP channels.
Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.
A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade. With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer. Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.
* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.
* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.
* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.
George Joseph [Fri, 3 Oct 2014 15:55:57 +0000 (15:55 +0000)]
sorcery: Prevent SEGV in sorcery_wizard_create when there's no create function
When you call ast_sorcery_create() you don't necessarily know which wizard is
going to be invoked. If it happens to be a wizard like 'config' that doesn't
have a 'create' virtual function you get a segfault in the
sorcery_wizard_create callback. This patch catches the null function pointer,
does an ast_assert, and logs an error.
Kinsey Moore [Fri, 3 Oct 2014 13:59:09 +0000 (13:59 +0000)]
PJSIP: Restore functional default for callerid_privacy
The pjsip config option default fixups from r424263 altered the
functional default from "allowed_not_screened" to "allowed". This
change restores the functional default value when none is provided.
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Kinsey Moore [Fri, 3 Oct 2014 13:33:11 +0000 (13:33 +0000)]
Manager: Add missing fields and documentation for CoreShowChannels
This corrects some issues introduced in the responses to the
CoreShowChannels AMI command as well as adding documentation for the
responses. The command in Asterisk 12 was missing the following fields:
Duration, Application, ApplicationData, and BridgedChannel and
BridgedUniqueID (replaced with BridgeId).
Richard Mudgett [Thu, 2 Oct 2014 21:55:37 +0000 (21:55 +0000)]
res_pjsip: Make transport cipher option accept a comma separated list of cipher names.
Improvements to the res_pjsip transport cipher option.
* Made the cipher option accept a comma separated list of OpenSSL cipher
names. Users of realtime will be glad if they have more than one name to
list.
* Added the CLI command 'pjsip list ciphers' so a user can know what
OpenSSL names are available for the cipher option.
* Updated the cipher option online XML documentation to specify what is
expected for the value.
* Updated pjsip.conf.sample to not indicate that ALL is acceptable since
ALL does not imply a preference order for the ciphers and PJSIP does not
simply pass the string to OpenSSL for interpretation.
Jonathan Rose [Thu, 2 Oct 2014 20:23:38 +0000 (20:23 +0000)]
Alembic: Add enumerator value to sippeers -> directmedia - 'outgoing'
The 'outgoing' value was left off of the enumerator when first creating the
column. This patch adds it, and should gracefully upgrade keeping the existing
data in tact.
ASTERISK-23781 #close
Reported by: Stephen More
Review: https://reviewboard.asterisk.org/r/4013/
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res_pjsip: document use of rewrite_contact in sample conf
Without setting rewrite_contact, an invite to an endpoint
behind NAT will not reach it - unless the endpoint itself
uses STUN or TURN to discover it's public URI. Thus, the
use of this should be in the sample documentation.
Joshua Colp [Wed, 1 Oct 2014 16:39:45 +0000 (16:39 +0000)]
res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.
During the latest update to DTLS-SRTP support the ability to configure
the hash used for fingerprints was added. This gave us two supported ones:
SHA-1 and SHA-256. The default was accordingly updated to SHA-256.
Unfortunately this configuration ability was not exposed within res_pjsip.
This change adds a dtls_fingerprint option that controls it.
#SIPit31
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Kinsey Moore [Wed, 1 Oct 2014 12:28:05 +0000 (12:28 +0000)]
PJSIP: Handle defaults properly
This updates the code behind PJSIP configuration options with custom
handlers to deal with the assigned default values properly where it
makes sense and adjusting the default value where it doesn't. Before
applying this patch, there were several cases where the default value
for an option would prevent that config section from loading properly.
Reported by: Thomas Thompson
Review: https://reviewboard.asterisk.org/r/4019/
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Kinsey Moore [Wed, 1 Oct 2014 12:15:56 +0000 (12:15 +0000)]
PJSIP: Force transport on contact rewrite
If contact rewriting is enabled but the contact differs in transport
from what is actually being used, messages after the initial INVITE
transaction can be sent to an incorrect transport/port combination. In
the case where this bug occurred the remote party never received a BYE
since it was sent to the remote party's TCP port over UDP.
res_rtp_asterisk: Ensure that the base and mapped address for candidates is present in SDP.
This change fixes an issue where ICE candidates put into the SDP did not contain
the 'raddr' and 'rport' information for server reflexive and relay candidates.
#SIPit31
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Richard Mudgett [Mon, 29 Sep 2014 21:18:54 +0000 (21:18 +0000)]
Simplify UUID generation in several places.
Replace code using ast_uuid_generate() with simpler and faster code using
ast_uuid_generate_str(). The new code avoids a malloc(), free(), and
copy.
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res_pjsip_session: Reduce SDP size by removing duplicate connection lines.
Due to the architecture of how media streams are handled each individual
handler adds connection details (IP address) for it. The first media stream
is then used as the top level SDP connection line. In practice each
line ends up being the same so to reduce the SDP size stream-level connection
information is also added to the SDP if it differs from the top level SDP
connection line.
res_pjsip_session: Add additional checks for delaying session refreshes.
There are certain situations which no checks existed for which need to prevent
session refreshes. This includes sending a session refresh with SDP before SDP
negotiation has completed and sending a session refresh before the dialog itself
has been established. Checks for these have been added.
Additionally COLP related UPDATEs were including SDP when it is not needed.
Richard Mudgett [Fri, 26 Sep 2014 15:28:39 +0000 (15:28 +0000)]
res_fax: Fix out of bounds error in update_modem_bits().
ASTERISK-24357 #close
Reported by: Jeremy Laine
Patches:
res_fax_bounds.patch (license #6561) patch uploaded by Jeremy Laine
Modified patch to not use magic numbers.
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Walter Doekes [Fri, 26 Sep 2014 14:41:38 +0000 (14:41 +0000)]
core: Don't allow free to mean ast_free (and malloc, etc..).
This gets rid of most old libc free/malloc/realloc and replaces them
with ast_free and friends. When compiling with MALLOC_DEBUG you'll
notice it when you're mistakenly using one of the libc variants. For
the legacy cases you can define WRAP_LIBC_MALLOC before including
asterisk.h.
Even better would be if the errors were also enabled when compiling
without MALLOC_DEBUG, but that's a slightly more invasive header
file change.
Those compiling addons/format_mp3 will need to rerun
./contrib/scripts/get_mp3_source.sh.
Richard Mudgett [Thu, 25 Sep 2014 21:03:51 +0000 (21:03 +0000)]
res_pjsip.c: Add missing off nominal cleanup in ast_sip_push_task_synchronous().
* Made memset the std struct in ast_sip_push_task_synchronous() because if
DEBUG_THREADS is enabled then uninitialized lock tracking data is used.
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Walter Doekes [Thu, 25 Sep 2014 20:49:04 +0000 (20:49 +0000)]
musiconhold: Add preferchannelclass=no option to prefer app class.
The new option 'preferchannelclass' is added to musiconhold.conf. If yes
(the default) the CHANNEL(musicclass) is preferred when choosing the
hold music. If it is no, the class suggested by the application that
calls the MoH (e.g. the Queue() app) gets preferred (new behaviour).
This way you set a different hold-music from the Queue-music by setting
both the CHANNEL(musicclass) and the queue-context musicclass.
ASTERISK-24276 #close
Reported by: Kristian Høgh
Patches:
app_override_channel_moh.patch uploaded by Kristian Høgh (License #6639)
Richard Mudgett [Wed, 24 Sep 2014 18:35:47 +0000 (18:35 +0000)]
pjsip_options.c: Fix race condition stopping periodic out of dialog OPTIONS request.
The crash on the issues is a result of an invalid transport configuration
change when asterisk is restarted. The attempt to send the qualify
request fails and we cleaned up. However, the callback is also called
which results in a double unref of the objects involved.
* Put a wrapper around pjsip_endpt_send_request() to detect when the
passed in callback is called because of an error so callers can know to
not cleanup.
* Made send_request_cb() able to handle repeated challenges (Up to 10).
* Fix periodic endpoint qualify OPTIONS sched deletion race by avoiding
it. The sched entry will no longer self stop and must be externally
stopped.
* Added REF_DEBUG description tags to struct sched_data in
pjsip_options.c.
* Fix some off-nominal ref leaks in schedule_qualify(),
qualify_and_schedule().
* Reordered pjsip_options.c module start/stop code to cleanup better on
error.
Mark Michelson [Tue, 23 Sep 2014 14:36:00 +0000 (14:36 +0000)]
Make CDR and CEL unit tests less FRACKy.
Prior to this commit, CDR and CEL tests were expected to trigger
FRACKs (i.e. assertions) due to the fact that the channels they
create have no formats on them. Some code was independently added
recently that attempts to prevent FRACKs from occurring by failing
early when attempting to set up translation paths if one or both
channels support no formats. Unfortunately, this attempt to be helpful
made the CDR and CEL tests go from simply FRACKing to outright
failing and in some cases, failing so badly as to crash Asterisk.
This commit seeks to correct past mistakes by adding the ulaw format
to channels created by the CDR and CEL unit tests. This makes setting
up translation paths succeed, eliminates previously-seen FRACKs, and
ultimately causes the unit tests to succeed again.
Matthew Jordan [Sun, 21 Sep 2014 01:16:05 +0000 (01:16 +0000)]
main/channel: Unlock channel in off-nominal path
In r423414 (13) / r423415 (trunk), an API call that determines if a format
capability structure is empty was added. This returns true if the format
capability structure is completely empty or "none". A check for this was added
in channel.c's set_format call. Unfortunately, when this check was true, it
returned from the function while still holding the channel lock. This caused
the CDR unit tests - which have a tendency to create channels with no formats -
to deadlock. Whoops.
This patch unlocks the channel on the off-nominal path.
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Prior to the release of Swagger 1.2, the attribute 'extends' was being
promoted as a possible way to show that a particular object extends an existing
object. Instead, the Swagger specification went with the 'subTypes' attribute
in the base object. This patch removes the unsupported attribute; the object
that the offending objects proposed to extend already lists them in its
'subTypes' attribute.
Matthew Jordan [Sat, 20 Sep 2014 23:41:55 +0000 (23:41 +0000)]
rest-api/api-docs: Correct basePath in resources to match top resources file
The resources.json file that defines the resource JSON files used with ARI
references a basePath of 'http://localhost:8088/ari'. This does not match what
is defined in the resource files themselves, 'http://localhost:8088/stasis'.
The correct base path is the one that includes 'ari' in the URL; this patch
updates the various resource JSON files to have the correct basePath.
res_pjsip_notify: Fix crash on unload/load and don't say the module doesn't exist on reload.
When unloading the module did not unregister the CLI commands causing a crash upon
load when they were registered again.
When reloading the module the return value from the config options framework was not
checked to determine if an error occurred or not. This caused a message to be output
saying the module did not exist when reloading if no changes were present.
AST-1433 #close
AST-1434 #close
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Richard Mudgett [Fri, 19 Sep 2014 17:16:32 +0000 (17:16 +0000)]
res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated.
Outgoing PJSIP calls can result in non-negotiated formats listed in the
channel's native formats if video formats are listed in the endpoint's
configuration. The resulting call could then use a non-negotiated format
resulting in one way audio.
* Simplified the update of session->req_caps in set_caps(). Why do
something in five steps when only one is needed?
Jonathan Rose [Fri, 19 Sep 2014 15:54:20 +0000 (15:54 +0000)]
Stasis_channels: Resolve unfinished Dials when doing masquerades
Masquerades into channels that are in the dialing state don't end their dial
and this goes against the model for things like CDRs and generating Dial end
manager actions and such.
ASTERISK-24237 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3990/
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Kinsey Moore [Fri, 19 Sep 2014 12:50:08 +0000 (12:50 +0000)]
PJSIP: Prevent T38 framehook being put on wrong channel
This change gives framehooks a reverse-direction masquerade callback in
addition to chan_fixup_cb similar to the callback added to datastores
to handle the same situation. The new callback provides the same
parameters as the fixup callback, but is called on the new channel's
framehooks before moving framehooks from the old channel to the new
channel. This gives the framehooks an oppurtunity to decide whether
they should remain on the new channel or be removed.
This new callback is used to prevent the PJSIP T.38 framehook from
remaining on a masqueraded channel if the new channel is not also a
PJSIP channel. This was causing a crash when a local channel was
masqueraded into a PJSIP channel and the framehook was executed on the
local channel since the channel's tech private data was not structured
as expected.
George Joseph [Thu, 18 Sep 2014 19:23:39 +0000 (19:23 +0000)]
utils: Create ast_strsep function that ignores separators inside quotes
This function acts like strsep with three exceptions...
* The separator is a single character instead of a string.
* Separators inside quotes are treated literally instead of like separators.
* You can elect to have leading and trailing whitespace and quotes
stripped from the result and have '\' sequences unescaped.
Like strsep, ast_strsep maintains no internal state and you can call it
recursively using different separators on the same storage.
Also like strsep, for consistent results, consecutive separators are not
collapsed so you may get an empty string as a valid result.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3989/
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Richard Mudgett [Thu, 18 Sep 2014 16:56:40 +0000 (16:56 +0000)]
astobj2.c/refcounter.py: Fix to deal with invalid object refs.
* Make astob2 REF_DEBUG output an invalid object line when an invalid ao2
object ref/unref is attempted. This is similar to the
constructor/destructor lines.
* Fixed refcounter.py to handle skewed objects that have
constructor/destructor states.
* Made refcounter.py highlight the invalid ao2 object refs by putting them
in their own section of the processed output file.
* Made refcounter.py highlight unreffing an object by more than one that
results in a negative ref count and the object being destroyed. The
abnormally destroyed object is reported in the invalid and finalized
object sections of the output.
Mark Michelson [Thu, 18 Sep 2014 16:38:26 +0000 (16:38 +0000)]
Add API call to determine if format capability structure is "empty".
Empty here means that there are no formats in the format_cap structure
or the only format in it is the "none" format.
I've added calls to check the emptiness of a format_cap in a few places
in order to short-circuit operations that would otherwise be pointless
as well as to prevent some assertions from being triggered in cases
where channels with no formats are used.
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Mark Michelson [Thu, 18 Sep 2014 16:09:25 +0000 (16:09 +0000)]
res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies.
res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE
arrives.
* It checks that there is a subscription handler for the Event
* It checks that there are body generators for the types in the Accept header
The problem is, there's nothing that ensures that these two things will
actually mesh with each other. For instance, Asterisk will accept a subscription
to MWI that accepts pidf+xml bodies. That doesn't make sense.
With this commit, we add some type information to the mix. Subscription
handlers state they generate data of type X, and body generators state
that they consume data of type X. This way, Asterisk doesn't end up in
some hilariously mismatched situation like the one in the previous paragraph.
George Joseph [Thu, 18 Sep 2014 15:14:38 +0000 (15:14 +0000)]
res_pjsip: ami: Fix error in AMI output when an endpoint has no transport
When no transport is associated to an endpoint, the AMI output for
PJSIPShowEndpoint indicates an error instead of silently ignoring the
missing transport.
This patch causes the error to appear only if a transport was specified
on the endpoint and the transport doesn't exist. It also fixes an issue
with counting the objects that were actually found.
ASTERISK-24161 #close
ASTERISK-24331 #close
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3998/
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David M. Lee [Thu, 18 Sep 2014 15:07:12 +0000 (15:07 +0000)]
Only install dahdi_span_config_hook if DAHDI is enabled
This patch changes the install to only install the hook script if
DAHDI is enabled. It also adds the script to the uninstall task, and
moves the DAHDI_UDEV_HOOK_DIR variable so that it's not between the
_MAKEOPTS variables and their comment.
This allows installs which specify a --prefix to work normally, as
long as they don't enable DAHDI.
George Joseph [Thu, 18 Sep 2014 14:46:12 +0000 (14:46 +0000)]
config: bug: Fix SEGV in ast_category_insert when matching category isn't found
If you call ast_category_insert with a match category that doesn't exist, the
list traverse runs out of 'next' categories and you get a SEGV. This patch
adds check for the end-of-list condition and changes the signature to return
an int for success/failure indication instead of a void.
The only consumer of this function is manager and it was also changed to use
the return value.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3993/
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projects / thirdparty / asterisk.git / log
