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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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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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
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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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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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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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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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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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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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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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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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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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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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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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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George Joseph
Matthew Jordan
Jonathan Rose
Joshua Colp
Richard Mudgett
Kinsey Moore
Igor Goncharovskiy
Alexandr Anikin
Corey Farrell
Walter Doekes
Kevin Harwell