asterisk/asterisk.git
5 years agores_pjsip: incorrect qualify statistics after disabling for contact
Kevin Harwell [Thu, 30 Oct 2014 17:18:47 +0000 (17:18 +0000)]
res_pjsip: incorrect qualify statistics after disabling for contact

When removing the qualify_frequency from an AoR or a contact the statistics
shown when issuing "pjsip show aors" from the CLI are incorrect. This patch
deletes the contact's status object from sorcery, disassociating it from the
contact, if the qualify_freqency is removed from configuration.

ASTERISK-24462 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4116/
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5 years agoapp_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE.
Walter Doekes [Thu, 30 Oct 2014 09:21:42 +0000 (09:21 +0000)]
app_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE.

In update_messages_by_imapuser(), messages were appended to a finite
array which resulted in a crash when an IMAP mailbox contained more
than 256 entries. This memory is now dynamically increased as needed.

Observe that this patch adds a bunch of XXX's to questionable code. See
the review (url below) for more information.

ASTERISK-24190 #close
Reported by: Nick Adams
Tested by: Nick Adams

Review: https://reviewboard.asterisk.org/r/4126/
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5 years agoAdd additional checks for NULL pointers to fix several crashes reported.
Igor Goncharovskiy [Thu, 30 Oct 2014 06:15:14 +0000 (06:15 +0000)]
Add additional checks for NULL pointers to fix several crashes reported.

ASTERISK-24304 #close
Reported by: dhanapathy sathya
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5 years agochannels/chan_sip: Add improved support for 4xx error codes
Matthew Jordan [Thu, 30 Oct 2014 01:59:39 +0000 (01:59 +0000)]
channels/chan_sip: Add improved support for 4xx error codes

This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER
response handling. This helps interoperability in a number of scenarios.

Review: https://reviewboard.asterisk.org/r/3437

patches:
  rb3437.patch uploaded by oej (License 5267)
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5 years agochannels/chan_sip: Support mutltiple Supported and Required headers
Matthew Jordan [Thu, 30 Oct 2014 01:48:00 +0000 (01:48 +0000)]
channels/chan_sip: Support mutltiple Supported and Required headers

A SIP request may contain multiple Supported: and Required: headers. Currently,
chan_sip only parses the first Supported/Required header it finds. This patch
adds support for multiple Supported/Required headers for INVITE requests.

Review: https://reviewboard.asterisk.org/r/2478

ASTERISK-21721 #close
Reported by: Olle Johansson
patches:
  rb2478.patch uploaded by oej (License 5267)
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5 years agoFix building chan_phone on big endian systems
Tzafrir Cohen [Wed, 29 Oct 2014 13:02:27 +0000 (13:02 +0000)]
Fix building chan_phone on big endian systems

A left over from the formats conversion (Corey Farrell).

ASTERISK-24458 #close
Review: https://reviewboard.asterisk.org/r/4117/

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5 years agobridge_builtin_features: Add missing channel locks around ast_get_chan_features_gener...
Richard Mudgett [Tue, 28 Oct 2014 21:35:41 +0000 (21:35 +0000)]
bridge_builtin_features: Add missing channel locks around ast_get_chan_features_general_config().

The feature_automonitor() and feature_automixmonitor() functions were not
locking the channel around ast_get_chan_features_general_config().
Accessing the channel datastore list without the channel locked is a good
way to corrupt the list or follow the pointer chain into oblivion.
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5 years agores_fax: Resolve T38 gateway frame leak.
Corey Farrell [Tue, 28 Oct 2014 21:10:42 +0000 (21:10 +0000)]
res_fax: Resolve T38 gateway frame leak.

When frames are translated by a fax gateway they need to be freed.  The
existing call to ast_frfree was unreachable.  This change reorganizes
fax_gateway_framehook to ensure that ast_frfree is called when needed.

ASTERISK-24457 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4115/
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5 years agomanager: Unsubscribe from acl_change_sub at shutdown.
Corey Farrell [Tue, 28 Oct 2014 20:44:22 +0000 (20:44 +0000)]
manager: Unsubscribe from acl_change_sub at shutdown.

ASTERISK-24453 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4110/
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5 years agoASTERISK-23512, correct inaccurate comment in manager.conf.sample
Malcolm Davenport [Tue, 28 Oct 2014 18:09:32 +0000 (18:09 +0000)]
ASTERISK-23512, correct inaccurate comment in manager.conf.sample

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agomain/bridge: Destroy features struct on off nominal path during bridge impart
Matthew Jordan [Tue, 28 Oct 2014 16:41:17 +0000 (16:41 +0000)]
main/bridge: Destroy features struct on off nominal path during bridge impart

When a channel is imparted to a bridge, the invocation of the function may
provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart,
the caller must assume that ownership has passed to the function, as in all
paths the function destroys the struct prior to returning (as its purpose is
to configure the behavior of the channel while in the bridge). On one off
nominal path - where the channel already has a PBX thread - the struct was not
being destroyed.

This patch fixes that glitch.

ASTERISK-24437 #close
Reported by: Scott Griepentrog
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5 years agomain/manager: Fix typo in AMI event documentation of "OriginateResponse"
Matthew Jordan [Tue, 28 Oct 2014 14:59:47 +0000 (14:59 +0000)]
main/manager: Fix typo in AMI event documentation of "OriginateResponse"

The parameter name is "Response", not "Resonse".

ASTERISK-24430 #close
Reported by: Dafi Ni
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5 years agoASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROL
Malcolm Davenport [Tue, 28 Oct 2014 14:57:01 +0000 (14:57 +0000)]
ASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROL

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5 years agoASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample
Malcolm Davenport [Tue, 28 Oct 2014 13:13:16 +0000 (13:13 +0000)]
ASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample

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5 years agoapp_queue: Cleanup ao2_iterator
Corey Farrell [Tue, 28 Oct 2014 11:22:55 +0000 (11:22 +0000)]
app_queue: Cleanup ao2_iterator

Clean ao2_iterator, resolving reference leak to queue members.

ASTERISK-24454 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4111/
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5 years agofunc_cdr: Fix CDR_PROP payload leak
Corey Farrell [Tue, 28 Oct 2014 11:12:03 +0000 (11:12 +0000)]
func_cdr: Fix CDR_PROP payload leak

Remove duplicate allocation of payload, preventing leak.

ASTERISK-24455 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4113/
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5 years agoconfigure: Add autoconf check for libopus.
Sean Bright [Mon, 27 Oct 2014 17:55:43 +0000 (17:55 +0000)]
configure: Add autoconf check for libopus.

Because opus transcoding support cannot be included in the standard Asterisk
distribution, a few codec_opus implementations have popped up.  To make it
easier for people to drop in opus support in their own installations, this
patch adds configure checks for libopus.

Review: https://reviewboard.asterisk.org/r/4106/
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5 years agores/res_http_websocket: Fix minor nits found by wdoekes on r409681
Matthew Jordan [Mon, 27 Oct 2014 02:47:03 +0000 (02:47 +0000)]
res/res_http_websocket: Fix minor nits found by wdoekes on r409681

When Moises committed the fixes for WSS (which was a great patch), wdoekes had
a few style nits that were on the review that got missed. This patch resolves
what I *think* were all of the ones that were still on the review.

Thanks to both moy for the patch, and wdoekes for the reviews.

Review: https://reviewboard.asterisk.org/r/3248/
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5 years agores/res_phoneprov: Fix crash on shutdown caused by container cleanup
Matthew Jordan [Mon, 27 Oct 2014 02:27:56 +0000 (02:27 +0000)]
res/res_phoneprov: Fix crash on shutdown caused by container cleanup

In res_phoneprov, unloading the module first destroys the http_routes
container, followed by the users. However, users may have a route in
the http_routes container; the validity of this container is not checked
in the users destructor. Hence, we hit an assert as the container has already
been set to NULL.

This patch does two things:
(1) It adds a sanity check in the user destructor (because why not)
(2) It switches the order of destruction, so that users are disposed of prior
    to the HTTP routes they may hold a reference to.

Note that this crash was caught by the Test Suite (go go testing!)
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5 years agores/res_srtp: Fix include issue for libsrtp 1.5.0
Matthew Jordan [Mon, 27 Oct 2014 01:47:56 +0000 (01:47 +0000)]
res/res_srtp: Fix include issue for libsrtp 1.5.0

In libsrtp 1.5.0, crypto_get_random is no longer resolved simply by including
srtp.h. Now, one must include crypto_kernel.h as well. As it turns out, this
header file has been provided by the library since 2006, so this is a
relatively benign change.

ASTERISK-24436 #close
Reported by: Patrick Laimbock
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5 years agoDocumentation: Improve documentation for ExtensionStatus AMI events
Jonathan Rose [Fri, 24 Oct 2014 15:32:35 +0000 (15:32 +0000)]
Documentation: Improve documentation for ExtensionStatus AMI events

Review: https://reviewboard.asterisk.org/r/4085/
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5 years agocodec_dahdi: Cannot use struct ast_translator.core_{src,src}_codec.
Shaun Ruffell [Wed, 22 Oct 2014 21:41:31 +0000 (21:41 +0000)]
codec_dahdi: Cannot use struct ast_translator.core_{src,src}_codec.

This fixes a Segmentation fault introduced in r419044 "media formats: re-architect
handling of media for performance improvements".

The problem is that codec_dahdi was using core_src_codec and core_dst_codec in the
ast_translator structure when these fields were never set. Now instead of trying to map
the new core codec descriptions to the way DAHDI defines different codecs, we will store
the DAHDI specific formats in 'struct translator' directly so we can refer to them without
mapping.

This also allows us to remove the "global_format_map" structure, since we can now query
the list of translators directly to make sure we do not ever register a DAHDI based
translator for a specific path more than once and eliminate the need to keep the list and
the map in sync.

ASTERISK-24435 #close
Reported by: Marian Koniuszko

Review: https://reviewboard.asterisk.org/r/4105/
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5 years agotranslage.c: Fix regression when generating translation path strings.
Richard Mudgett [Tue, 21 Oct 2014 18:04:43 +0000 (18:04 +0000)]
translage.c: Fix regression when generating translation path strings.

Fix the AMI Status action read and write translation path strings from
growing for each channel in the status event list by reseting the ast
string given to ast_translate_path_to_str() to fill in the given
translation path.
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5 years agoAST-2014-011: Fix POODLE security issues
Matthew Jordan [Mon, 20 Oct 2014 14:20:15 +0000 (14:20 +0000)]
AST-2014-011: Fix POODLE security issues

There are two aspects to the vulnerability:
(1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use
    TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the
    TCP/TLS core, which should be done as an improvement at a latter date.
(2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified,
    will default to the OpenSSL SSLv23_method. This method allows for all
    ecnryption methods, including SSLv2/SSLv3. A MITM can exploit this by
    forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE.
    This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration,
    and explicitly disables SSLv2/SSLv3 if using SSLv23_method.

For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is
explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or
SSLv3.

Much thanks to abelbeck for reporting the vulnerability and providing a patch
for the res_jabber/res_xmpp modules.

Review: https://reviewboard.asterisk.org/r/4096/

ASTERISK-24425 #close
Reported by: abelbeck
Tested by: abelbeck, opsmonitor, gtjoseph
patches:
  asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903)
  asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903)
  AST-2014-011-1.8.diff uploaded by mjordan (License 6283)
  AST-2014-011-11.diff uploaded by mjordan (License 6283)
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5 years agobuild: Force -fsigned-char on platforms where the default for char is unsigned
George Joseph [Sun, 19 Oct 2014 17:09:38 +0000 (17:09 +0000)]
build: Force -fsigned-char on platforms where the default for char is unsigned

gcc on the ARM platform defaults 'char' to 'unsigned char' whereas Intel and
SPARC default to 'signed char'.  This is only an issue in the rare cases where
negative values are assigned to a 'char' but this this patch insures
compatibility by detecting platforms that default to 'unsigned' and adding an
'-fsigned-char' flag to _ASTCFLAGS.

If compiling for ARM (native or cross-compile) be sure to run ./bootstrap.sh
and ./configure to regenerate the build files.  You shouldn't have to do this
for Intel or SPARC.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4091/
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5 years agores/res_pjsip_sdp_rtp: Revert 425924
Matthew Jordan [Sun, 19 Oct 2014 04:03:35 +0000 (04:03 +0000)]
res/res_pjsip_sdp_rtp: Revert 425924

This patch for r425924 introduced a bug, wherein sending an INVITE request
with no SDP would cause Asterisk to not send an SDP Offer in the 200
OK. The current structure of res_pjsip_sdp_rtp is a bit hard to deal with
to fix this, as create_outgoing_sdp has no knowledge of whether or not it is
creating an SDP as a new Offer or an Answer. This is something of an oversight
in the callback definition, as the caller of it does have this information.

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5 years agores/res_pjsip_sdp_rtp: Remove left over reference to override_prefs
Matthew Jordan [Sun, 19 Oct 2014 00:56:43 +0000 (00:56 +0000)]
res/res_pjsip_sdp_rtp: Remove left over reference to override_prefs

The usage of the local override_prefs variable in create_outgoing_sdp_stream
was previously to track an override format preference set by PJSIP_MEDIA_OFFER.
Now, however, that function simply sets the joint capabilities structure,
session->req_caps. During the media format rework, the override_prefs was
instead used to check if there were any formats in session->req_caps.

However, this usage isn't useful in create_outgoing_sdp_stream.
session->req_caps contains the negotiated formats for *all* streams, not just
the current one being created. Thus, so long as any stream of any type has
provided a format, override_prefs will be non-zero. Hence, its usage in
checking whether or not we should look at the formats on the endpoint or
the joint capabilities is generally useless.

There's only two things useful to check:
(1) Does the endpoint have a format for the media type?
(2) Did we negotiate a format for the media type?

If either of those is a 'no', then we must kill the media stream.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425924 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoSample Configurations: make 'pjsip reload' reload all reloadable pjsip modules
Jonathan Rose [Fri, 17 Oct 2014 22:45:27 +0000 (22:45 +0000)]
Sample Configurations: make 'pjsip reload' reload all reloadable pjsip modules

AST-1432 #close
Reported by: John Bigelow
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5 years agores_pjsip_session/res_pjsip_sdp_rtp: Be more tolerant of offers
Matthew Jordan [Fri, 17 Oct 2014 13:35:44 +0000 (13:35 +0000)]
res_pjsip_session/res_pjsip_sdp_rtp: Be more tolerant of offers

When an inbound SDP offer is received, Asterisk currently makes a few
incorrection assumptions:

(1) If the offer contains more than a single audio/video stream, Asterisk will
    reject the entire stream with a 488. This is an overly strict response;
    generally, Asterisk should accept the media streams that it can accept and
    decline the others.
(2) If the offer contains a declined media stream, Asterisk will attempt to
    process it anyway. This can result in attempting to match format
    capabilities on a declined media stream, leading to a 488. Asterisk should
    simply ignore declined media streams.
(3) Asterisk will currently attempt to handle offers with AVPF with
    use_avpf=No/AVP with use_avpf=Yes. This mismatch results in invalid SDP
    answers being sent in response. If there is a mismatch between the media
    type being offered and the configuration, Asterisk must reject the offer
    with a 488.

This patch does the following:
* Asterisk will accept SDP offers with at least one media stream that it can
  use. Some WARNING messages have been dropped to NOTICEs as a result.
* Asterisk will not accept an offer with a media type that doesn't match its
  configuration.
* Asterisk will ignore declined media streams properly.

#SIPit31

Review: https://reviewboard.asterisk.org/r/4063/

ASTERISK-24122 #close
Reported by: James Van Vleet

ASTERISK-24381 #close
Reported by: Matt Jordan
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5 years agores_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriente...
Joshua Colp [Fri, 17 Oct 2014 13:17:58 +0000 (13:17 +0000)]
res_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriented transports.

This change adds a module which is configurable using the keep_alive_interval setting in the
global section that will send a CRLF keep alive to all active connection-oriented transports at
the provided interval. This is useful because it can help keep connections open through NATs.
This functionality also exists within PJSIP but can not be controlled at runtime and requires
recompiling it.

Review: https://reviewboard.asterisk.org/r/4084/

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5 years agochannels/chan_sip: Respect outboundproxy setting when sending qualify requests
Matthew Jordan [Fri, 17 Oct 2014 13:11:07 +0000 (13:11 +0000)]
channels/chan_sip: Respect outboundproxy setting when sending qualify requests

The outboundproxy setting is currently ignored when sending OPTIONS requests
as a result of the qualify setting. This means that if an Asterisk server is
unable to send the packet directly to a peer, it is unable to qualify any
non-inbound registered peer (e.g. a peer SIP Trunk).

This patch grabs the outboundproxy information for a peer when a qualify
attempt is being constructed and, if it finds the information, uses it
when sending the OPTIONS request.

Review: https://reviewboard.asterisk.org/r/3948

ASTERISK-24063 #close
Reported by: Damian Ivereigh
patches:
  outboundproxy-dai.patch uploaded by Damian Ivereigh (License 6632)
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5 years agores_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable.
Joshua Colp [Fri, 17 Oct 2014 11:30:23 +0000 (11:30 +0000)]
res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable.

This change adds a configuration option which adds a 'user=phone' parameter if the user
portion of the request URI or the From URI is determined to be a number.

Review: https://reviewboard.asterisk.org/r/4073/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425804 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoAMI: Add missing VarSet events when a channel inherits variables.
Richard Mudgett [Fri, 17 Oct 2014 02:49:57 +0000 (02:49 +0000)]
AMI: Add missing VarSet events when a channel inherits variables.

There should be AMI VarSet events when channel variables are inherited by
an outgoing channel.  Also local;2 should generate VarSet events when it
gets all of its channel variables from channel local;1.

ASTERISK-24415 #close
Reported by: Richard Mudgett
Patches:
      jira_asterisk_24415_v12.patch (license #5621) patch uploaded by Richard Mudgett

Review: https://reviewboard.asterisk.org/r/4074/
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5 years agobridge_native_rtp: Fix audio issues when moving from remote bridge to softmix
Matthew Jordan [Fri, 17 Oct 2014 02:01:40 +0000 (02:01 +0000)]
bridge_native_rtp: Fix audio issues when moving from remote bridge to softmix

When a native RTP bridge that is remotely bridging its participants switches
to a softmix bridge, it may not properly re-INVITE the media for one or both
participants back to Asterisk. This is due to the current bridge_native_rtp
code only re-INVITEs if it believes the channel will survive the bridge
operation. Currently, that code is failing, as it expects the channels to
have a soft hangup flag set on it indicating that a redirect has occurred
or that the channel is going to leave the bridge. (The code did not take into
account a smart bridge operation).

This patch also renames a few things to be more reflective of the underlying
types.

Review: https://reviewboard.asterisk.org/r/3997/

ASTERISK-24327 #close
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5 years agotest_cel: Update pickup test to expect CANCEL instead of ANSWSER
Matthew Jordan [Fri, 17 Oct 2014 01:46:07 +0000 (01:46 +0000)]
test_cel: Update pickup test to expect CANCEL instead of ANSWSER

The CEL pickup test previously looked for a disposition of ANSWER between the
original caller/peer when the call is picked up. This is actually incorrect:
the disposition should, at the very least, not be ANSWER as the call was
never ANSWERed. The disposition is now CANCEL; this patch updates the test
accordingly.
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5 years agomain/cdr: Use 'time' when rescheduling batched CDRs as opposed to 'size'
Matthew Jordan [Thu, 16 Oct 2014 21:21:44 +0000 (21:21 +0000)]
main/cdr: Use 'time' when rescheduling batched CDRs as opposed to 'size'

When refactoring CDRs to use the configuration framework, a 'whoops' was
introduced where the CDR batch size was used when rescheduling a batch,
as opposed to the time duration. This patch corrects that obvious mistake.

ASTERISK-24426 #close
Reported by: Shane Blaser
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5 years agoconfig: Fix inf loop using ast_category_browse and ast_variable_retrieve
George Joseph [Thu, 16 Oct 2014 17:32:16 +0000 (17:32 +0000)]
config: Fix inf loop using ast_category_browse and ast_variable_retrieve

Fix infinite loop when calling ast_variable_retrieve inside an
ast_category_browse loop when there is more than 1 category with
the same name.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4089/
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5 years agoPJSIP: Enforce module load dependencies
Kinsey Moore [Thu, 16 Oct 2014 16:32:25 +0000 (16:32 +0000)]
PJSIP: Enforce module load dependencies

This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub
have loaded properly before attempting to load any modules that depend
on them since the module loader system is not currently capable of
resolving module dependencies on its own.

ASTERISK-24312 #close
Reported by: Dafi Ni
Review: https://reviewboard.asterisk.org/r/4062/
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5 years agoFix loss of voice after second call drops (on a second line) in case using multiple...
Igor Goncharovskiy [Thu, 16 Oct 2014 06:22:07 +0000 (06:22 +0000)]
Fix loss of voice after second call drops (on a second line) in case using multiple lines on unistim phones. There is regression was introduced in r391379.

Reported by: Rustam Khankishyiev
(closes issue ASTERISK-23846)
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5 years agores_rtp_asterisk: Fix a bug where ICE state would get reset when it shouldn't.
Joshua Colp [Thu, 16 Oct 2014 01:26:18 +0000 (01:26 +0000)]
res_rtp_asterisk: Fix a bug where ICE state would get reset when it shouldn't.

In the case where the ICE negotiation had not yet started current state would
get wiped when it shouldn't.

This also removes channel binding as in practice this does not work well with
other implementations.
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5 years agochan_motif: Cleanup jingle_tech.capabilities only once.
Richard Mudgett [Wed, 15 Oct 2014 19:39:15 +0000 (19:39 +0000)]
chan_motif: Cleanup jingle_tech.capabilities only once.
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5 years agoparking_tests: Fix assertions and possibly crashes in res_parking unit tests
Jonathan Rose [Wed, 15 Oct 2014 19:17:29 +0000 (19:17 +0000)]
parking_tests: Fix assertions and possibly crashes in res_parking unit tests

Assertions were caused by attempting to play music on hold to a channel with
no formats. Parking unit test channels were given formats and a technology so
that they would be able to pretend to read/write frames.

ASTERISK-24413 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4075/
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5 years agochan_ooh323: fix rtptimeout general value checking
Alexandr Anikin [Wed, 15 Oct 2014 10:03:05 +0000 (10:03 +0000)]
chan_ooh323: fix rtptimeout general value checking

correct condition to check rtptimeout in [general] config section

ASTERISK-24393 #close
Reported by:  Dmitry Melekhov
Tested by:  Dmitry Melekhov
Patches:
  ASTERISK-24393.patch
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5 years agoconfig: Fix SEGV in unit test with MALLOC_DEBUG
George Joseph [Tue, 14 Oct 2014 20:48:06 +0000 (20:48 +0000)]
config: Fix SEGV in unit test with MALLOC_DEBUG

With MALLOC_DEBUG the /main/config config_basic_ops test was causing a
SEGV while doing an ast_category_delete in an ast_category_browse loop.
Apparently this never worked but was also never tested.  I removed the
test, added 2 notes to config.h indicating that it's not supported and
added a few lines of code to ast_category_delete to prevent the SEGV
should someone attempt it in the future.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4078/
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5 years agoScheduler: Fix a nasty scheduler caching bug which makes new tasks not execute
Jonathan Rose [Tue, 14 Oct 2014 19:12:58 +0000 (19:12 +0000)]
Scheduler: Fix a nasty scheduler caching bug which makes new tasks not execute

Tasks that were marked for pending deletion in the scheduler would be moved to
the cache for later reuse, but after being recycled the deleted mark wouldn't
be removed resulting in fresh tasks being deleted without reason... and
immediately moved back into the cache where they could be reused again. This
could cause horrendous things to happen in just about anything that used a
scheduler.

ASTERISK-24321 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4071/
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5 years agores_phoneprov: Create accessor for ast_phoneprov_std_variable_lookup
George Joseph [Tue, 14 Oct 2014 18:13:33 +0000 (18:13 +0000)]
res_phoneprov: Create accessor for ast_phoneprov_std_variable_lookup

Based on feedback from Richard, I created an accessor for
res_phoneprov/ast_phoneprov_std_variable_lookup and added
load priority to AST_MODULE_INFO.

Tested-by: George Joseph
Tested-by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/4076/
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5 years agores_fax: Fix reference leak caused by gateway sessions
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.

ASTERISK-24392 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4049/
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5 years agostasis_channels.c: Resolve unfinished Dials when doing masquerades (Part 2)
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

Review: https://reviewboard.asterisk.org/r/4066/
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5 years agores_fax: Resolve module reference leak caused by reserved sessions
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.

ASTERISK-18923 #close
Reported by: Grigoriy Puzankin
Review: https://reviewboard.asterisk.org/r/4048/
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5 years agomanager/config: Support templates and non-unique category names via AMI
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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5 years agores_rtp_asterisk: Make the ICE transport check case insensitive as some implementatio...
Joshua Colp [Sun, 12 Oct 2014 21:09:49 +0000 (21:09 +0000)]
res_rtp_asterisk: Make the ICE transport check case insensitive as some implementations use 'udp'.
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5 years agochan_sip: Fix so asterisk won't send reINVITE after a BYE.
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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5 years agobuild: Relax badshell tilde test to allow for ~ in middle of DESTDIR.
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.

ASTERISK-13797 #close
Reported by: Tzafrir Cohen

Review: https://reviewboard.asterisk.org/r/4064/
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5 years agores_calendar_ews: Relax neon version check to work with 0.30 too.
Walter Doekes [Sun, 12 Oct 2014 07:47:52 +0000 (07:47 +0000)]
res_calendar_ews: Relax neon version check to work with 0.30 too.

Allow res_calendar_ews to work not only with libneon-0.29 but also
with 0.30.

ASTERISK-24325 #close
Reported by: Tzafrir Cohen

Review: https://reviewboard.asterisk.org/r/4068/
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5 years agores_phoneprov: Cleanup module load error handling
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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5 years agobridge: During a smart bridge operation provide a more complete bridge to the old...
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.

Review: https://reviewboard.asterisk.org/r/4057/
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5 years agores/res_phoneprov: Bail on registration if res_phoneprov didn't load
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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5 years agores_pjsip_phoneprov_provider: Add missing dependency on pjproject.
Joshua Colp [Fri, 10 Oct 2014 14:24:57 +0000 (14:24 +0000)]
res_pjsip_phoneprov_provider: Add missing dependency on pjproject.
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5 years agoCallerID: Fix parsing regression
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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5 years agores_pjsip_nat: Place source port into rport of responses if 'force_rport' is on.
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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5 years agochan_sip: Fix dialog leak resulting from missing ACK to re-INVITE.
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.

(Patch reindented by me.)

ASTERISK-20784 #close
ASTERISK-15879 #close
Reported by: Torrey Searle, Nitesh Bansal
Patches:
  reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334)
  patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418)

Reviewboard: https://reviewboard.asterisk.org/r/4052/
(testcase can be found at r4051)
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5 years agores_pjsip_phoneprov_provider: fix compile breakage on AST_VECTOR
George Joseph [Thu, 9 Oct 2014 23:37:49 +0000 (23:37 +0000)]
res_pjsip_phoneprov_provider: fix compile breakage on AST_VECTOR

endpoint->inbound_auths was changed to a vector in 13 and I
committed the 12 patch instead of the 13 patch.

Tested-by: George Joseph
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5 years agores_rtp_asterisk: Crash if no candidates received for component
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.

ASTERISK-24383 #close
Review: https://reviewboard.asterisk.org/r/4039/
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5 years agores_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov
George Joseph [Thu, 9 Oct 2014 20:55:50 +0000 (20:55 +0000)]
res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov

This module allows res_pjsip to integrate with res_phoneprov.  It handles
the pjsip 'phoneprov' object type.

Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/3976/
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5 years agores/res_phoneprov: Don't cancel Asterisk load on module load failure
Matthew Jordan [Thu, 9 Oct 2014 18:44:00 +0000 (18:44 +0000)]
res/res_phoneprov: Don't cancel Asterisk load on module load failure
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5 years agores_phoneprov: Refactor phoneprov to allow pluggable config providers
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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5 years agocdr.c: Make turning on CDR debug a one step process instead of two.
Richard Mudgett [Thu, 9 Oct 2014 16:38:40 +0000 (16:38 +0000)]
cdr.c: Make turning on CDR debug a one step process instead of two.

Now "cdr set debug on" doesn't also require "core set verbose 1" to see
CDR debug output.
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5 years agosafe_asterisk: Don't automatically exceed MAXFILES value of 2^20.
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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5 years agores_rtp_asterisk: Allow only UDP ICE candidates.
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.

ASTERISK-24326 #close
Reported by: Joshua Colp
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5 years agoStasis: Relegate log message to dev-mode
Kinsey Moore [Wed, 8 Oct 2014 18:24:47 +0000 (18:24 +0000)]
Stasis: Relegate log message to dev-mode

This error message primarily applies to development tasks and will now
only show up when dev-mode is enabled via configure.
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5 years agoIndexer: Format message types may not exist
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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5 years agoStasis: Only log errors for non-declined types
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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5 years agodata: Properly access formats in capabilities structure when adding codecs.
Joshua Colp [Tue, 7 Oct 2014 18:34:40 +0000 (18:34 +0000)]
data: Properly access formats in capabilities structure when adding codecs.

Formats within a capabilities structure are addressed starting at 0, not 1.
Assuming 1 causes it to exceed an array.

ASTERISK-24389 #close
Reported by: Kevin Harwell
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5 years agores/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameter
Matthew Jordan [Tue, 7 Oct 2014 17:44:36 +0000 (17:44 +0000)]
res/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameter

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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5 years agores/res_pjsip_pubsub: Fix typo in WARNING message
Matthew Jordan [Tue, 7 Oct 2014 14:09:47 +0000 (14:09 +0000)]
res/res_pjsip_pubsub: Fix typo in WARNING message
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5 years agomessage: Don't close an AMI connection on SendMessage action error
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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5 years agofeatures.c: Fix lingering channel ref while Bridge() application is active.
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.

* Eliminated bridge_exec()'s use of RAII_VAR().

ASTERISK-24224 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/4041/
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5 years agosdp_srtp: Add new lines to some WARNING messages
Matthew Jordan [Mon, 6 Oct 2014 12:39:03 +0000 (12:39 +0000)]
sdp_srtp: Add new lines to some WARNING messages
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5 years agores_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpoint
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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5 years agopjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels
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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5 years agores_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message
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:

(1) res_pjsip_logger
(2) res_hep_pjsip
(3) res_pjsip/distributor

Anything that can intercept an rdata object before res_pjsip/distributor should
be defensive when looking at the received packet.

#SIPit31

ASTERISK-24369 #close
Reported by: Matt Jordan
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5 years agores/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions
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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5 years agoRelease AMI connections on shutdown.
Corey Farrell [Sun, 5 Oct 2014 00:49:45 +0000 (00:49 +0000)]
Release AMI connections on shutdown.

ASTERISK-24378 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4037/
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5 years agochan_motif: Correct last commit to use ao2_cleanup to free format cap
Corey Farrell [Sun, 5 Oct 2014 00:15:43 +0000 (00:15 +0000)]
chan_motif: Correct last commit to use ao2_cleanup to free format cap

This fix applies to 13 and trunk.

ASTERISK-24384 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4043/
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5 years agochan_motif: Release format capabilities and config on module load error
Corey Farrell [Sun, 5 Oct 2014 00:02:39 +0000 (00:02 +0000)]
chan_motif: Release format capabilities and config on module load error

ASTERISK-24384 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4043/
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5 years agores_pjsip: Fix XML typo and update CHANGES.
Richard Mudgett [Fri, 3 Oct 2014 21:58:03 +0000 (21:58 +0000)]
res_pjsip: Fix XML typo and update CHANGES.

ASTERISK-24199
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5 years agoaudiohooks: Reevaluate the bridge technology when an audiohook is added or removed.
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.

ASTERISK-24195 #close
Reported by: Jonathan Rose

Review: https://reviewboard.asterisk.org/r/4046/
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5 years agoapp_queue: Add dialplan function to get the channel name at the specified position...
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.

Review: https://reviewboard.asterisk.org/r/4035/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424493 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_pjsip: Fix deadlock when masquerading PJSIP channels.
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.

ASTERISK-24356 #close
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/4034/
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5 years agosorcery: Prevent SEGV in sorcery_wizard_create when there's no create function
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.

Review: https://reviewboard.asterisk.org/r/4044/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424449 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoPJSIP: Restore functional default for callerid_privacy
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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5 years agoManager: Add missing fields and documentation for CoreShowChannels
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).

ASTERISK-24262 #close
Reported by: Mitch Claborn
Review: https://reviewboard.asterisk.org/r/4040/
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5 years agores_pjsip: Make transport cipher option accept a comma separated list of cipher names.
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.

ASTERISK-24199 #close
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/4018/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424395 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoAlembic: Add enumerator value to sippeers -> directmedia - 'outgoing'
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424380 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_pjsip: Fix an assertion for channels that lack formats on creation
Jonathan Rose [Thu, 2 Oct 2014 15:33:50 +0000 (15:33 +0000)]
chan_pjsip: Fix an assertion for channels that lack formats on creation

ASTERISK-24222 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4017/
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5 years agores_pjsip: document use of rewrite_contact in sample conf
Scott Griepentrog [Thu, 2 Oct 2014 13:36:01 +0000 (13:36 +0000)]
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.

Review: https://reviewboard.asterisk.org/r/4036/
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Merged revisions 424338 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424339 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_hep: Release allocation reference to configuration.
Corey Farrell [Wed, 1 Oct 2014 20:37:31 +0000 (20:37 +0000)]
res_hep: Release allocation reference to configuration.

ASTERISK-24362 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4026/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.
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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Merged revisions 424290 from http://svn.asterisk.org/svn/asterisk/branches/12
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5 years agores_pjsip_sdp_rtp: Accept DTLS attributes in top level, not just media session.
Joshua Colp [Wed, 1 Oct 2014 16:20:40 +0000 (16:20 +0000)]
res_pjsip_sdp_rtp: Accept DTLS attributes in top level, not just media session.

#SIPit31
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Merged revisions 424287 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 424288 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424289 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoPJSIP: Handle defaults properly
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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