api.wiki.mustache: Update wiki template to support body parameters

This patch updates the api.wiki.mustache template and the swagger_model python
script to understand if an operation has a body parameter. If an operation
does have a body parameter, it will now be displayed in the corresponding
wiki entry.
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Merged revisions 407389 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@417877 f38db490-d61c-443f-a65b-d21fe96a405b
ARI: Improvements to body parameters documentation

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/
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@417879 f38db490-d61c-443f-a65b-d21fe96a405b
main/tcptls: Add checks for OpenSSL Elliptic Curve support

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: http://svn.asterisk.org/svn/asterisk/trunk@417900 f38db490-d61c-443f-a65b-d21fe96a405b
HTTP: Add persistent connection support.

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/
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Merged revisions 417880 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@417901 f38db490-d61c-443f-a65b-d21fe96a405b
pbx_config: Add manager actions to add/remove extensions

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: http://svn.asterisk.org/svn/asterisk/trunk@417910 f38db490-d61c-443f-a65b-d21fe96a405b
chan_dahdi: Add AMI commands for controlling PRI debugging output

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: http://svn.asterisk.org/svn/asterisk/trunk@417916 f38db490-d61c-443f-a65b-d21fe96a405b
res_ari: Fix some off-nominal paths just dropping the HTTP connection.

* 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
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Merged revisions 417932 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@417933 f38db490-d61c-443f-a65b-d21fe96a405b
chan_dahdi: Add inband_on_setup_ack compatibility option.

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/
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Merged revisions 417956 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 417957 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 417958 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@417976 f38db490-d61c-443f-a65b-d21fe96a405b
Remove many deprecated modules

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: http://svn.asterisk.org/svn/asterisk/trunk@418019 f38db490-d61c-443f-a65b-d21fe96a405b
configure: Update script to pass if channels/h323/Makefile.in does not exist

This simply removes that check from the configure script, as r418019 removed
chan_h323.


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@418034 f38db490-d61c-443f-a65b-d21fe96a405b
configure: Remove last vestiges of h323; DO create menuselect-deps

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: http://svn.asterisk.org/svn/asterisk/trunk@418035 f38db490-d61c-443f-a65b-d21fe96a405b
main/Makefile: fix compilation error of buildinfo occurring on 'make install'

Egads. Another bad deletion of too much when attempting to remove h323 stuff.



git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@418050 f38db490-d61c-443f-a65b-d21fe96a405b
HTTP: Fix build for gcc 4.10
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Merged revisions 418066 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@418067 f38db490-d61c-443f-a65b-d21fe96a405b
CEL: Fix incorrect/missing extra field information

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/
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Merged revisions 418071 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@418084 f38db490-d61c-443f-a65b-d21fe96a405b
ARI/res_stasis: Subscribe to both Local channel halves when originating to app

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
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Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@418090 f38db490-d61c-443f-a65b-d21fe96a405b
res_pjsip_dialog_info_body_generator: Add dialog-info+xml support for presence.

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/
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Merged revisions 418116 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@418117 f38db490-d61c-443f-a65b-d21fe96a405b