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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418507 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Install a hook script for DAHDI to register new spans with Asterisk
automatically by running:

  asterisk -rx 'dahdi create channel FIRST LAST'

Review: https://reviewboard.asterisk.org/r/3157/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417863 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP.

This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide
a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation
completes. Configuration options to chan_sip and chan_pjsip have also been added to
allow behavior to be tweaked (such as forcing the AVP type media transports in SDP).

ASTERISK-22961 #close
Reported by: Jay Jideliov

Review: https://reviewboard.asterisk.org/r/3679/
Review: https://reviewboard.asterisk.org/r/3686/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417679 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When processing a 212MB refs file, refcounter.py used over 3GB of RAM.
This change greatly reduces memory usage in two ways:

* Saving object history in whole lines instead of separated values.
* Not saving normal/skewed/leaked object lists unless they are requested.

ASTERISK-23921 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3668/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417485 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change makes res_pjsip_pubsub persist inbound subscriptions in sorcery. By default
this uses the local astdb but it can also be configured to store within an outside
database. When Asterisk is started these subscriptions are recreated if they have not
expired. Notifications are sent to the devices which have subscribed and they are none
the wiser that the system has restarted.

Review: https://reviewboard.asterisk.org/r/3598/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415767 65c4cc65-6c06-0410-ace0-fbb531ad65f3

From now on, make install will overwrite safe_asterisk with the
latest version. You need to move any local modifications to files
inside /etc/asterisk/startup.d, if you have any.

See also commits r394939 and r397938.

ASTERISK-21965 #close
Patches:
  safe_asterisk.patch uploaded by jkister (License 6232, modified by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Replaced a stray echo that should've been a message call in
  safe_asterisk. This replaces a conditional log message by a slightly
  different message. Please update your log parsing scripts.
* Made the $NOTIFY mail Subject more verbose by adding the machine name
  and exitstatus.

(Note that a 'make install' still won't overwrite your old safe_asterisk
if it exists. See ASTERISK-21965.)

ASTERISK-23492 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415524 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Cleans up the safe_asterisk script and adds the ASTSAFE_FOREGROUND
option that allows the debian asterisk init script to capture the
right pid.

* Drop the vim #modeline which wasn't used. Use test consistently
  without the odd configure xno syntax. Double quote all paths.
  General cleanup.
* Don't output message()s to the console but only to TTY if set.
* Allow TTY to be "no" as well as empty (debian compatibility with
  debian/patches/safe_asterisk-config).
* Add option to export ASTSAFE_FOREGROUND=1 from the init script
  that calls this to disable backgrounding. Debian uses a similar
  method in debian/patches/safe_asterisk-nobg).

ASTERISK-23492 #close
Review: https://reviewboard.asterisk.org/r/3574/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415173 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When generating SQL files via the repotools alembic_creator.py script, a
configuration object is used programatically with SQLAlechemy, as opposed to
a configuration file. This patch ignores failures to interpret a config file,
as ... there isn't one in this case.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The string lengths on certain columns created through alembic for PJSIP were
too short. For instance, columns containing URIs are currently set to 40
characters, but this can be too small and result in truncated values.  Added
an alembic migration script that increases the size of these columns and a
few others to 255.

ASTERISK-23639 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3475/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412993 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
    REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
    Every run will now blow away the previous run (as large ref files
    sometimes caused issues). We now also no longer open/close the file
    on each write, instead relying on fflush to make sure data gets written
    to the file (in case the ao2 call being performed is about to cause a
    crash)
(3) It goes with a comma delineated format for the ref debug file. This
    makes parsing much easier. This also now includes the thread ID of the
    thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
    contrib/scripts folder.
(5) The old refcounter implementation in utils/ has been removed.

Review: https://reviewboard.asterisk.org/r/3377/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412154 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since the relatime scripts are now managed by Alembic, the previous realtime
scripts were previously removed. However, the removal process messed up, as
the files were still in the repository. The contents were just empty.

This removes the files from the tree.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411443 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-23235)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3324/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410575 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Transport TOS values were interpreted as DSCP values without being documented
as such. Endpoint TOS values (tos_audio/tos_video) behaved normally as TOS
values have historically. This patch makes the transport TOS values behave as
TOS values and makes all TOS values readable as string values (e.g. AF11).
In addition, alembic scripts have been updated to use the proper field types
for all TOS/COS values.

(issue ASTERISK-23235)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3304/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410029 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Added the queues and queue_members tables to the config alembic scripts.

* Added the CDR table alembic creation script.  The CDR table is more of
an example for new setups since the actual table can be fully customized
in cdr_adaptive_odbc.conf.

(closes issue ASTERISK-23233)
Reported by: jmls

Review: https://reviewboard.asterisk.org/r/3227/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408299 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If an enum had been previously created the alembic script would attempt to
re-create it and an error would be generated while running migrations for a
postgresql server.  The work around for this is to use the ENUM object type
for postgres as opposed to the generic enum type used by sqlalchemy. Using
this type in the script seems to work properly for both postgres and mysql.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407574 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added a "debug" configuration option for res_pjsip that when set to "yes"
enables SIP messages to be logged.  It is specified under the "system" type.
Also added an alembic script to add the option to realtime.

(closes issue ASTERISK-23038)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3148/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407037 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A couple of the scripts had errors that would not allow a full migration to
take place.  The extensions table needed to make its 'id' column a primary
key in order to work with mysql.  The other script ...add_endpoints... was
missing tables that it was trying to add columns to.

Added the primary key on id for extensions and added the tables in for the
missing pjsip configuration options.  While it is not ideal to modify already
released scripts this was a case where it had to be done due to errors in
the script and lacking a better alternative.

Review: https://reviewboard.asterisk.org/r/3167/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407029 65c4cc65-6c06-0410-ace0-fbb531ad65f3

live_ast can be used as a wrapper script to run asterisk, gdb or
valgrind. In those cases it runs them and returns the result. It is more
useful to use 'exec' to avoid having another odd process in the chain.

Review: https://reviewboard.asterisk.org/r/3110/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406503 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds Path support to chan_pjsip in res_pjsip_path.c with minimal
additions in res_pjsip_registrar.c to store the path and additions in
res_pjsip_outbound_registration.c to enable advertisement of path
support to registrars and intervening proxies.

Path information is stored on contacts and is enabled via Address of
Record (AoRs) and Registration configuration sections.

While adding path support, it became necessary to be able to add SIP
supplements that handled messages outside of sessions, so a framework
for handling these types of hooks was added in parallel to the
already-existing session supplements and several senders of
out-of-dialog requests were refactored as a result.

(closes issue ASTERISK-21084)
Review: https://reviewboard.asterisk.org/r/3050/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the alembic scripts were written for creating Asterisk
realtime databases the extensions table for dialplan wasn't
included.  This update creates the extensions table.

(closes issue ASTERISK-22815)
Reported by: Zone Conkle
Review: https://reviewboard.asterisk.org/r/3064/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403714 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Updated the alembic script for pjsip.  Also, the dtls config parsing stuff was
expecting strings with no underscores, so removed the underscores from the
option name before passing it to the parser.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403083 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Renamed, where appropriate, the configuration options for chan/res_pjsip to use
snake case (compound words separated by an underscore).  For example, faxdetect
will become fax_detect, recordofffeature will become record_off_feature, etc...

Review: https://reviewboard.asterisk.org/r/3002/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403051 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-22374)
Reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2846
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402328 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
1) The env scripts have been updated to be tolerant of a NULL configuration
   file. This occurs when configuration is provided by an external script,
   such that the actual config.ini file is not used.
2) Enum types have all been given names. This is needed for PostgreSQL script
   generation.
3) The identifier meetme_confno_starttime_endtime is greater than 30
   characters, and hence invalid for Oracle databases. This has been truncated
   down to meetme_confno_start_end.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402073 65c4cc65-6c06-0410-ace0-fbb531ad65f3

so that it can be served by the Asterisk web server.

I had to change the Makefile in order to recursively install content
from the static-http directory, hence the code review instead of just
putting it in.

Review: https://reviewboard.asterisk.org/r/2924/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401263 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also updates sample configurations for sorcery and extconfig to
demonstrate how to use databases created by that alembic script.

(closes issue ASTERISK-22133)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2892/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400533 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter
and Tzafrir have pointed out numerous issues with the approach and have
propsed an alternative in r/2757. Since it's not a time critical issue and
is not worth holding up the release of 12 for it, I've gone ahead and reverted
r394939 from 12/trunk and re-opened ASTERISK-21965.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In r397874, the scripts were removed... but not replaced. Thanks to
Michael Young for noticing this!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397912 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch replaces contrib/realtime/ with a new setup for managing the
database schema required for database integration with Asterisk.  In
addition to initializing a database with the proper schema, alembic can do a
database migration to assist with upgrading Asterisk in the future.
Hopefully this helps make setting up and operating Asterisk with a database
easier.

With this the schema only needs to be maintained in one place instead of
once per database.  The schemas I have added here have a bit of improvement
over the examples that were there before (some added consistency and added
some missing indexes).  Managing the schema in one place here also applies
to all databases supported by SQLAlchemy.

See contrib/ast-db-manage/README.md for more details.

Review: https://reviewboard.asterisk.org/r/2731

patch by Russell Bryant (license 6300)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397875 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Most, if not all, of the backing features of a conf file should now be
implemented (e.g. multi-line comments, includes, templates, etc...).  A
few of the options still need to be mapped.  Those are currently listed
in the 'sip_to_res_sip.py' file.

Things to do:

(1) There is more work to do here, at least for the sip.conf items that
aren't currently parsed. An issue will be created for that.
(2) All of the scripts should probably be passed through pylint and have
as many PEP8 issues fixed as possible.
(3) A public review is probably warranted at that point of the entire script.

Reported by: Matt Jordan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.

Review: https://reviewboard.asterisk.org/r/2708/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also fixes spacing, in passing.

(closes issue ASTERISK-22131)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395283 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch modifies the behavior of safe_asterisk in two ways:
(1) It modifies the Asterisk Makefile such that safe_asterisk is always
    installed on a 'make install'. This was done as bugfixes in the
    safe_asterisk script were not applied in previous version of Asterisk
    without first removing the old version of the script.
(2) In order to keep a newly installed version of safe_asterisk from impacting
    local modifications, a new config file - safe_asterisk.conf.sample - has
    been provided. Settings that were previously modified in safe_asterisk can
    be set there instead.

(closes issue ASTERISK-21965)
Reported by: Jeremy Kister
patches:
  safe_asterisk.patch uploaded by jkister (License 6232)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch updates the realtime SQL scripts with an entry that will create the
queue_log table. This brings the PostgreSQL scripts inline with the MySQL
scripts, with respect to what tables they will create.

(closes issue ASTERISK-21021)
Reported by: Eugene
patches:
  queue_log.sql uploaded by varnav (license 6360)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394898 65c4cc65-6c06-0410-ace0-fbb531ad65f3

** This script is in no way finished.

Started the initial "cut" at converting a sip.conf file to a res_sip.conf file.
Hopefully the bulk of the framework is in place and only a few minor adjustments
need to be made when an option mapping is added that "doesn't fit".  This script
and supporting files should be executable against python version 2.5.

An OrderedDict class (backported from a newer version of python) is included.
A MultiOrderedDict class is implemented so options, when added, should be able
to be added in order and allowed to have multiple values.

Currently the scripts supports the majority of endpoint options found in
res_sip.conf.  Support has also been added for Aor(s) and the ACL/security
sections.  Inside the sip_to_res_sip.py file one can see a list of options
that still need to be mapped.

Also items that still need to be done: templates, includes, parsing '=>'
delimiter.  Note that some code is hopefully in place already to support
templates (e.g. lookup/retrieving defaults from them).  However, the
parsing of and adding of the section needs to be done.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394024 65c4cc65-6c06-0410-ace0-fbb531ad65f3