Age | Commit message (Collapse) | Author |
|
Whenever possible, audiohooks and framehooks will now be copied over
to the channel that the masquerading channel gets cloned into. This
should occur for all audiohooks and most framehooks. As a result,
in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
deprecated and its behavior is essentially the new default for all
audiohooks, plus some additional audiohooks/framehooks.
Review: https://reviewboard.asterisk.org/r/3721/
........
Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Specifically the following equivalents were created:
fax show session -> FAXSession
fax show sessions -> FAXSessions
fax show stats -> FAXStats
Review: https://reviewboard.asterisk.org/r/3666/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.
Review: https://reviewboard.asterisk.org/r/3804/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
progressinband=never in sip.conf is easily defeated if an onward trunk sends a
progress indication of its own. This is almost certain to happen if the onward
trunk is ISDN or IAX as these technologies send a progress indication even if
early media is not required. This progress message is passed to the caller,
and causes the "never" option to be rather badly named.
This patch changes the behaviour of this setting in the following ways:
1) In sip_write(), do not pass the media unless we have either progressed
beyond INV_EARLY_MEDIA, or we are in INV_EARLY_MEDIA state, and early
media is both set-up and wanted. This helps resolve double-ringing on some
buggy handsets.
2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, but
SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to avoid implicitly
enabling early media. Avoid sending double ring indications.
NOTE: the meaning of the SIP_PROGRESS_SENT flag changes slightly in this patch
to also encapsulate the fact that a channel has *sent or received* a 183
Progress indication. This makes the updated code in sip_write() much more
simple.
Review: https://reviewboard.asterisk.org/r/3700
ASTERISK-23972 #close
Reported by: Steve Davies
patches:
inband_never_present_early_media2 uploaded by Steve Davies (License 5012)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The commit that added libxml2 support didn't fully check for the libxml2
development script in the Asterisk configure file. As a result, Asterisk could
be configured, then fail on menuselect. This patch fixes it so that Asterisk
should detect the libxml2 dependency failure first.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This is the final patch in adding menuselect to Asterisk.
- The first patch (r418832) added menuselect along with mxml
- The second patch (r418833) removed mxml from menuselect
This patch adds support for libxml2 to menuselect, and makes libxml2 a
required library for Asterisk.
Note that the libxml2 portion of this patch was written by Sean Bright,
and was made available on a team branch:
http://svn.digium.com/svn/menuselect/team/seanbright/libxml2/
Review: https://reviewboard.asterisk.org/r/3773/
ASTERISK-20703 #close
patches:
some_mysterious_team_branch uploaded by seanbright (License 5060)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This is the second patch that adds menuselect to Asterisk trunk. The previous
commit (r418832) added menuselect along with mxml; this patch removes mxml
completely from Menuselect.
A subsequent patch will switch menuselect over to using libxml2, and make
libxml2 a required dependency for Asterisk.
ASTERISK-20703
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This is the first patch that adds menuselect to Asterisk trunk, and removes
the svn:externals property. This is being done for two reasons:
(1) The removal of external repositories eases a future migration to git
(2) Asterisk is now the only thing that uses menuselect; as a result, there's
little need to keep it in an external repository
Subsequent patches will remove the mxml dependency from menuselect and tidy
up the build system.
ASTERISK-20703
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Ensure that three-way transfers can be reported even if featuremap is
non-NULL.
........
Merged revisions 418810 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The include was unneeded, this is split off from r3758 as it applies to 12.
........
Merged revisions 418787 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Most channel drivers let you specify a default accountcode to be set on
channels associated with a particular peer/endpoint/object. Prior to this
patch, chan_pjsip/res_pjsip did not support such a setting.
This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'.
When a channel is created that is associated with an endpoint with this value
set, the channel will automatically have its accountcode property set to the
value configured for the endpoint.
Review: https://reviewboard.asterisk.org/r/3724/
ASTERISK-24000 #close
Reported by: Matt Jordan
........
Merged revisions 418756 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch adds support for the PostgreSQL application_name connection setting.
When the appropriate PostgreSQL module's configuration is set with an
application name, the name will be passed to PostgreSQL on connection and
displayed in the database's pg_stat_activity view, as well as in CSV logs. This
aids in managing which applications/servers are connected to a PostgreSQL
database, as well as tracing the activity of those connections.
Review: https://reviewboard.asterisk.org/r/3591
ASTERISK-23737 #close
Reported by: Gergely Domodi
patches:
pgsql_application_name.patch uploaded by Gergely Domodi (License 6610)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Technically, ADPCM is a method that can be applied to several codecs.
Asterisk's ADPCM codec is the Dialogic ADPCM or VOX codec.
See http://en.wikipedia.org/wiki/Dialogic_ADPCM for more information
about said codec.
Review: https://reviewboard.asterisk.org/r/3744
patches:
rb3744.patch uploaded by dennis.guse (License 6513)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When the PresenceState action is executed, the nominal path fails to include
the ActionID in the successful response. This patch adds a call to
astman_start_ack, which guarantees that an ActionID (if provided) will be
sent back to the AMI client.
Unlike the Asterisk 11 and 12 patches, this patch also deprecates the
duplicate Message key in the response to the action, replacing it with the
key 'PresenceMessage'.
Review: https://reviewboard.asterisk.org/r/3776/
ASTERISK-23985 #close
........
Merged revisions 418713 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 418714 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This fixes two reference leaks that would occur when TEST_FRAMEWORK was
enabled and features were successfully executed.
........
Merged revisions 418715 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Previously these two dialplan functions would issue warnings and
return failure when an empty string is used as the argument. Now
they will not issue a warning and will successfully return an
empty string.
ASTERISK-23911 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3745/
........
Merged revisions 418641 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 418649 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 418650 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
It's been 2014 for like... 6 months.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 418586 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Update refcounter.py to use delta's to track the current reference count.
* Use result from internal_ao2_ref to write old_refcount to refs_log.
Review: https://reviewboard.asterisk.org/r/3756/
........
Merged revisions 418504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 418505 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 418506 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This change maps the ao2_t_cleanup() function to the
correct debug function so that it can be used.
Review: https://reviewboard.asterisk.org/r/3764/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Cleanup games object in app_skel.
* Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+).
Review: https://reviewboard.asterisk.org/r/3757/
........
Merged revisions 418465 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 418466 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This change removes h323.conf.sample, jingle.h, jabber.h left behind by r3698.
Review: https://reviewboard.asterisk.org/r/3755/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Tags are useful in hunting down ref imbalances; this patch adds tag variants
for these commonly used macros/functions.
Review: https://reviewboard.asterisk.org/r/3750/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This change causes ao2_replace to do nothing when src == dst. This
avoids REF_DEBUG logging when we're not actually doing anything.
Review: https://reviewboard.asterisk.org/r/3743/
........
Merged revisions 418396 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When updated configuration is written back to the conf
file - for example when a user changes their voicemail
pin, make sure that any config hook that wants to know
of changes is informed.
Review: https://reviewboard.asterisk.org/r/3708/
........
Merged revisions 418366 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 418369 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This is a whitespace only change.
........
Merged revisions 418323 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 418324 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
creation.
Square pegs in round holes don't work very well.
........
Merged revisions 418261 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 418262 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 418263 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Create a Stasis bridge sub-class to propagate linkedids and
accountcodes.
* Fixed the basic bridge sub-class to update peeraccount codes when the
number of channels in the bridge drops back down to two parties.
* Refactored ast_bridge_channel_update_accountcodes() to handle channels
joining/leaving the bridge.
* Fixed the basic bridge sub-class to not call the base bridge class pull
method twice.
AFS-105 #close
ASTERISK-23852 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3720/
........
Merged revisions 418225 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 418182 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The dtls_perform_handshake function was mistakenly placed under the guards for
USE_PJPROJECT. If PJPROJECT was not installed, the function would not be
defined, while other functions would attempt to still use it. This prevented
res_rtp_asterisk from being loaded.
ASTERISK-24001 #close
Reported by: Don Fanning
........
Merged revisions 418172 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This module implements dialog-info+xml for the purposes of presence. This means
that phones such as Grandstreams can now subscribe to receive presence information
for an extension.
ASTERISK-21443 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3705/
........
Merged revisions 418116 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch fixes two bugs:
1. When originating a channel into a Stasis application, we already create a
subscription for the channel that is going into our Stasis app.
Unfortunately, when you create a Local channel and pass it off to a Stasis
app, you really aren't creating just one channel: you're creating two. This
patch snags the second half of the Local channel pair (assuming it is a
Local channel pair, but luckily core_local is kind about such assumptions)
and subscribes to it as well.
2. Subscriptions are a bit sticky right now. If a subscription is made, the
'interest' count gets bumped on the Stasis subscription - but unless
something explicitly unsubscribes the channel, said subscription sticks
around. This is not much of a problem is a user is creating the subscription
- if they made it, they must want it. However, when we are creating
implicit subscriptions, we need to make sure something clears them out.
This patch takes a pessimistic approach: it watches the cache updates
coming from Stasis and, if we notice that the cache just cleared out an
object, we delete our subscription object. This keeps our ao2 container of
Stasis forwards in an application from growing out of hand; it also is a
bit more forgiving for end users who may not realize they were supposed to
unsubscribe from that channel that just hung up.
Review: https://reviewboard.asterisk.org/r/3710/
#ASTERISK-23939 #close
........
Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This corrects two issues with the extra field information in Asterisk
12+ in channel event logs.
It is possible to inject custom values into the dialstatus provided by
ast_channel_dial_type() Stasis messages that fall outside the
enumeration allowed for the DIALSTATUS channel variable. CEL now
filters for the allowed values and ignores other values.
The "hangupsource" extra field key is always blank if the far end
channel is a chan_pjsip channel. This is because the hangupsource is
never set for the pjsip channel driver. This change sets the
hangupsource whenever a hangup is queued for chan_pjsip channels.
This corrects an issue with the pjsip channel driver where the
hangupcause information was not being set properly.
Review: https://reviewboard.asterisk.org/r/3690/
........
Merged revisions 418071 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 418066 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Egads. Another bad deletion of too much when attempting to remove h323 stuff.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The previous patch (r418034) fixed the 'glitch' that the channels/h323
Makefile no longer existed. Unfortunately, removing the entire line was a bit
of a blunder, as it meant that build_tools/menuselect-deps was never
generated. Hilarity ensued when actually trying to compile.
But hey! At least configure worked.
This patch fixes *that* glitch, and removes some more of the vestiges of h323.
(It had tendrils in the main Makefile? Crazy.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This simply removes that check from the configure script, as r418019 removed
chan_h323.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Billing records are fair,
To get paid is quite bright,
You should really use ODBC;
Good-bye cdr_sqlite.
Microsoft did once push H.323,
Hell, we all remember NetMeeting.
But try to compile chan_h323 now
And you will take quite a beating.
The XMPP and SIP war was fierce,
And in the distant fray
Was birthed res_jabber/chan_jingle;
But neither to stay.
For everyone did care and chase what Google professed.
"Free Internet Calling" was what devotees cried,
But Google did change the specs so often
That the developers were happy the day chan_gtalk died.
And then there was that odd application
Dedicated to the Polish tongue.
app_saycountpl was subsumed by Say;
One could say its bell was rung.
To read and parse a file from the dialplan
You could (I guess) use an application.
app_readfile did fill that purpose, but I think
A function is perhaps better in its creation.
Barging is rude, I'm not sure why we do it.
Inwardly, the caller will probably sigh.
But if you really must do it,
Don't use app_dahdibarge, use ChanSpy.
We all despise the sound of tinny robots
It makes our queues so cold.
To control such an abomination
It's better to not use Wait/SetMusicOnHold.
It's often nice to know properties of a channel
It makes our calls right
We have a nice function called CHANNEL
And so SIPCHANINFO is sent off into the night.
And now things get odd;
Apparently one could delimit with a colon
Properties from the SIPPEER function!
Commas are in; all others are done.
Finally, a word on pipes and commas.
We're sorry. We can't say it enough.
But those compatibility options in asterisk.conf;
To maintain them forever was just too tough.
This patch removes:
* cdr_sqlite
* chan_gtalk
* chan_jingle
* chan_h323
* res_jabber
* app_saycountpl
* app_readfile
* app_dahdibarge
It removes the following applications/functions:
* WaitMusicOnHold
* SetMusicOnHold
* SIPCHANINFO
It removes the colon delimiter from the SIPPEER function.
Finally, it also removes all compatibility options that were configurable from
asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems.
Review: https://reviewboard.asterisk.org/r/3698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The new inband_on_setup_ack option causes Asterisk to assume inband audio
may be present when a SETUP_ACKNOWLEDGE message is received.
Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
dialtone is sent from the network side, progress indicator 8 "Inband info
now available" MAY be sent to the CPE if no digits were received with the
SETUP. It is thus implied that the ie is mandatory if digits came with
the SETUP and dialtone is needed. This option should be enabled, when the
network sends dialtone and you want to hear it, but the network doesn't
send the progress indicator when needed.
NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
dialing is also enabled because Q.SIG does not send the progress indicator
with the SETUP ACK.
The commit -r413714 (AST-1338) which causes this issue was dealing with a
SIP-to-ISDN interoperability issue.
This commit is a merge of the two patches indicated below.
ASTERISK-23897 #close
Reported by: Pavel Troller
Patches:
pri-4.diff (license #6302) patch uploaded by Pavel Troller
jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/3633/
........
Merged revisions 417956 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 417957 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 417958 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Removed some incorrect newlines on ast_http_error() messages in
manager.c.
* Removed an incorrect newline in res_ari_channels.c.
Addendum to ASTERISK-23552
........
Merged revisions 417932 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Adds the following AMI commands:
PRIDebugSet - Set PRI debug levels for a specific span
PRIDebugFileSet - Set the file used for PRI debug message output
PRIDebugFileUnset - Disables file output for PRI debug messages
Review: https://reviewboard.asterisk.org/r/3681/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Adds two new manager commands to pbx_config - DialplanExtensionAdd and
DialplanExtensionRemove which allow manager users to create and delete
extensions respectively.
Review: https://reviewboard.asterisk.org/r/3650/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Persistent HTTP connection support is needed due to the increased usage of
the Asterisk core HTTP transport and the frequency at which REST API calls
are going to be issued.
* Add http.conf session_keep_alive option to enable persistent
connections.
* Parse and discard optional chunked body extension information and
trailing request headers.
* Increased the maximum application/json and
application/x-www-form-urlencoded body size allowed to 4k. The previous
1k was kind of small.
* Removed a couple inlined versions of ast_http_manid_from_vars() by
calling the function. manager.c:generic_http_callback() and
res_http_post.c:http_post_callback()
* Add missing va_end() in ast_ari_response_error().
* Eliminated unnecessary RAII_VAR() use in http.c:auth_create().
ASTERISK-23552 #close
Reported by: Scott Griepentrog
Review: https://reviewboard.asterisk.org/r/3691/
........
Merged revisions 417880 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The patch for ASTERISK-23905 that added PFS support in Asterisk depends on the
elliptic curve library support being present in OpenSSL. As it turns out, some
versions of OpenSSL don't have this library - notably the version running on
our build agents.
This patch fixes the build by providing a configure check for the specific
library calls that the PFS patch relies on.
Review: https://reviewboard.asterisk.org/r/3709/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The variables body parameter under the originate and originate with id
operations of the channel resource showed invalid JSON in its description.
The variables body parameter under the userEvent operation of the event
resource made no mention that the custom key/value pairs should be wrapped
in a variables key in order to be added to the custom user event.
ASTERISK-23975 #close
Review: https://reviewboard.asterisk.org/r/3692/
........
Merged revisions 417878 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|