Joshua Colp [Tue, 4 Nov 2014 12:03:35 +0000 (12:03 +0000)]
res_pjsip_outbound_registration: Add virtual line support.
Virtual line support establishes a relationship between messages
related to an outbound registration and a local endpoint. This is
accomplished by attaching a parameter to the Contact of the outbound
registration and looking for it on any received requests. If the
parameter exists and can be matched to an outbound registration
the configured endpoint is associated with the request.
Richard Mudgett [Mon, 3 Nov 2014 18:22:59 +0000 (18:22 +0000)]
res_pjsip: Add disable_tcp_switch option.
When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In
some circumstances (on some networks), this can cause some issues with
messages not getting sent to the correct destination - and can also cause
connections to get dropped due to quirks in pjproject deciding to
terminate TCP connections with no messages.
While fixing the routing/messaging issues is important, having a
configuration option in Asterisk that tells pjproject to not switch over
to TCP would be useful. That way, if some glitch is discovered on some
other network/site, we can at least disable the behavior until a fix is
put into place.
Joshua Colp [Mon, 3 Nov 2014 14:45:01 +0000 (14:45 +0000)]
chan_pjsip: Add support for passing hold and unhold requests through.
This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.
Corey Farrell [Sun, 2 Nov 2014 08:13:52 +0000 (08:13 +0000)]
Fix ast_writestream leaks
Fix cleanup in __ast_play_and_record where others[x] may be leaked.
This was caught where prepend != NULL && outmsg != NULL, once
realfile[x] == NULL any further others[x] would be leaked. A cleanup
block was also added for prepend != NULL && outmsg == NULL.
11+: Fix leak of ast_writestream recording_fs in
app_voicemail:leave_voicemail.
Matthew Jordan [Sun, 2 Nov 2014 01:01:52 +0000 (01:01 +0000)]
res/res_stasis: Fix crash on module unload while performing operation
When the res_stasis module is unloaded, it will dispose of the apps_registry
container. This is a problem if an ARI operation is in flight that attempts
to use the registry, as the shutdown occurs in a separate thread. This patch
adds some sanity checks to the various routines that access the registry which
cause the operations to fail if the apps_registry does not exist.
Crash caught by the Asterisk Test Suite.
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A question arose as to whether a .pem file
could be provided in place of the .crt and
.key files in a PJSIP TLS configuration. I
tested this and discovered that although a
cert will be read from the pem file, a key
will not, and thus the priv_key_file entry
is still required. This update to the fine
documentation clarifies the option usage.
AST-1448 #close
Review: https://reviewboard.asterisk.org/r/4129/
Reported by: John Bigelow
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This updates the status of the outbound registration
to reflect when it has been unregistered. Since the
registration is unregistered but is not stopped, the
registration schedule remains active as before. The
patch also updates the documentation of both the AMI
and CLI commands.
ASTERISK-24411 #close
Review: https://reviewboard.asterisk.org/r/4119/
Reported by: John Bigelow
patches:
unregister-patch1.txt uploaded by John Bigelow (License 5091)
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Matthew Jordan [Fri, 31 Oct 2014 03:26:28 +0000 (03:26 +0000)]
channels/sip/reqresp_parser: Fix unit tests for r426594
When r426594 was made, it did not take into account a unit test that verified
that the function properly populated the unsupported buffer. The function
would previously memset the buffer if it detected it had any contents; since
this function can now be called iteratively on successive headers, the unit
tests would now fail. This patch updates the unit tests to reset the buffer
themselves between successive calls, and updates the documentation of the
function to note that this is now required.
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Corey Farrell [Thu, 30 Oct 2014 23:56:39 +0000 (23:56 +0000)]
app_queue: fix a couple leaks to struct call_queue in set_member_value
set_member_value has a couple leaks to references in the variable q
found through testsuite tests/queues/set_penalty. Also remove the
REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible
with the updated REF_DEBUG code.
Currently, it is possible for some subscriptions to get into a NULL state. When
this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a
device is subscribed for extension state then the associated subscription state
object can't be located. The code then attempts to dereference a NULL object.
Added a NULL check to avoid the problem.
Reported by: John Bigelow
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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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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
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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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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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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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.
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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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.
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.
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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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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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
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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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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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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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.
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.
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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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.
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.
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.
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
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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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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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.
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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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.
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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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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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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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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projects / thirdparty / asterisk.git / log
