This change implements SRV support for the IP based endpoint
identifier module. All possible addresses through SRV are looked
up and added as matches. If no SRV records are available a
fallback to normal host resolution is done. If an IP address
is provided then no SRV lookup occurs.

This is configured using the "srv_lookups" option on the
identify section and defaults to "yes".

ASTERISK-26693

Change-Id: I6b641e275bf96629320efa8b479737062aed82ac

Pyflake is a python (2) source checker. This patch fixes various
(mostly trivial) errors and warnings it reports.

Change-Id: Ia35c5ac61751b927814cf693994c632c412386ea

Change-Id: I8aa55a7c3fb175235ddc7f85e9457d5102d06fa7

When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.

The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.

The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.

ASTERISK-26423 #close

Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc

This change adds support for dual stack automatically. No
configuration is required and the IP address and version
in the SIP messages and SDP will be automatically changed
based on the transport over which the message is being
sent. RTP usage has also been changed to listen on both
IPv4 and IPv6 simultaneously to allow media to flow, and
to allow ICE support on both simultaneously. This also
allows failover between IPv6 and IPv4 to work as expected.

ASTERISK-26309 #close

Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d

cdr, config and voicemail are all separate alembic trees.  Because
alembic's default is to use a table named 'alembic_version' to store
the current tree revision, the 3 trees can't exist in the same schema
without stepping on each other.

Now each tree uses 'alembic_version_<tree_name>' as the version table.
Each tree's env.py script now first checks for 'alembic_version'.  If
it finds it AND its revision is in the tree's history, the script
renames it to 'alembic_version_<tree_name>'.  Regardless, the script
then continues with the migration using 'alembic_version_<tree_name>'
and creates that table if it's not found.  The result is that if an
existing 'alembic_version' table was found but it didn't belong to this
tree, it's left alone and 'alembic_version_<tree_name>' is used or
created.

WARNING:  If multiple trees are using the same schema, they MUST NOT
CRU or D any objects with names that might exist in the other trees.
An example would be 'yesno_values' type.  If two trees perform
operations on it, one tree could pull it out from under the other.
Thankfully we currently don't share any names among cdr, config and
voicemail.

NOTE:  Since the env.py scripts in each tree were identical, a common
env.py has been placed in the ast-db-manage directory and a symlink
to it has been placed in each tree directory.

ASTERISK-24311 #close
Reported-by: Dafi Ni

Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898

Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636

Map the sip.conf general section legacy_useroption_parsing to the
new pjsip.conf global ignore_uri_user_options.

ASTERISK-26316
Reported by: Kevin Harwell

Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc

This implements the chan_sip legacy_useroption_parsing option but with a
better name.

* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.

ASTERISK-26316 #close
Reported by: Kevin Harwell

Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62

If you use the safe_asterisk script, it uses hardcoded defaults before
running configurable values from /etc/asterisk/startup.d. The hardcoded
default has TTY=9. Some containerized environments don't have such a
TTY, and safe_asterisk would stop.

The custom configuration from /etc/asterisk/startup.d/* isn't read until
after it stopped, so changing TTY in a custom config did not help.

This changeset changes safe_asterisk to continue if the TTY setting was
untouched and /dev/tty9 and /dev/vc/9 aren't found.

Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc

This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.

ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn

Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762

Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2

Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15

Change-Id: If35174614545727817d329c60ba4456c028941b5

When using the migration script sip_to_pjsip.py, and your sip.conf is
configured with bindaddr=::, two transports are written to pjsip.conf, one for
0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4
and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface
like in chan_sip.

Furthermore, the script internal functions "build_host" and "split_hostport"
did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change
makes sure, even such addresses are parsed correctly.

ASTERISK-26309

Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48

When using the migration script sip_to_pjsip.py, cert_file was not migrated to
pjsip.conf. A previous change regarding this contained a copy/paste error.

ASTERISK-22374

Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b

A recent update had a copy/paste error where the unused variable 'val' was
being passed to the set_value function instead of the 'method' value itself.

This patch passes in the right variable.

ASTERISK-22374

Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06

When using the migration script sip_to_pjsip.py and tlsclientmethod is not set
in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to
overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is
offering/using not just TLSv1.0 but TLSv1.2 as well.

ASTERISK-22374

Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f

When using the migration script sip_to_pjsip.py, no section of type=system or
type=general were created. Therefore the keys compactheaders, timerb, timert1,
and useragent were not migrated to pjsip.conf.

ASTERISK-22374

Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1

When using the migration script sip_to_pjsip.py, session-timers=accept and
session-timers=refuse were mapped to wrong values.

ASTERISK-22374

Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092

When using the migration script sip_to_pjsip.py, now the (mandatory) username is
written to pjsip.conf, even if there was no (optional) authname in the register
string in sip.conf.

ASTERISK-22374

Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f

When using the migration script sip_to_pjsip.py and the register string
started with a transport in sip.conf - like tls://... - register was not parsed
correctly and therefore not migrated correctly to pjsip.conf.

ASTERISK-22374

Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2

When using the migration script sip_to_pjsip.py, those keys got missing. These
keys might appear several times and the function "merge_value" tried to collect
those. However, because these keys have different names in sip.conf and
pjsip.conf, "merge_value" was not able to find the new key name in sip.conf.
This change lets "merge_value" search with the old key name in sip.conf and
write with the new key name in pjsip.conf.

ASTERISK-22374

Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2

When using the migration script sip_to_pjsip.py, the externhost or externip of
sip.conf were erroneously written to Endpoints instead to Transports.

ASTERISK-22374

Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4

When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and
minexpiry were not migrated to pjsip.conf.

ASTERISK-22374

Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b

When using the migration script sip_to_pjsip.py, encryption=yes got missing and
media_encryption=sdes was not written to pjsip.conf, because of a typo.

ASTERISK-22374

Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05

When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got
missed, because of a typo. Therefore, cos and tos were not written to
pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused
by a copy-and-paste error.

ASTERISK-22374

Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2

When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were
not migrated to pjsip.conf.

ASTERISK-22374

Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825

contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.

Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4

A new identify_by option was added recently, auth_username. However, this
setting was not added as an allowable choice in the database enumeration
value.

This patch updates the current enumeration, adding in the new setting.

ASTERISK-26268 #close

Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8

The extensions table defined two columns (id and priority) as primary key
autoincrement columns. However only one is allowed when defining the primary
key.

This patch removes the autoincrement attribute from the priority column since
it does not need to be as such and really should not have been on there in the
first place.

This patch also removes 'context', 'exten', and 'priority' from the primary key
index and creates a new combined unique contraint index on them.

ASTERISK-26183 #close

Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b

The PJSIP taskprocessors could be overflowed on startup
if there are many (thousands) realtime endpoints
configured with unsolicited mwi.
The PJSIP stack could be totally unresponsive for a few minutes
after boot completed.

This patch creates a separate PJSIP serializers pool for mwi
and makes unsolicited mwi use serializers from this pool.
This patch also adds 2 new global options to tune taskprocessor
alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.

This patch also adds new global option 'mwi_disable_initial_unsolicited'
to disable sending unsolicited mwi to all endpoints on startup.
If disabled then unsolicited mwi will start processing
on next endpoint's contact update.

ASTERISK-26230 #close

Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a

The regular expression would match causing the code that handled
the line if it was merely a comment to never get executed.

Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819

Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5

When configuring SIP URIs in the pjsip.conf file it is
necessary to escape the semicolon so the parser does not
treat it as a comment. This change allows this to work in
the astconfigparser implementation.

A secondary bug where some data was lost if a configuration
option included a "=" in its value was also fixed.

A bug where sections would be considered equal despite
being different has also been fixed.

Change-Id: If229f656ef22050b50e7b34e90c4bffe796431f8

sqlalchemy was complaining:

sqlalchemy.exc.IdentifierError: Identifier
'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30
characters

This fixes the problem by changing the index name to be
"ps_contacts_qualifyfreq_exp" instead.

ASTERISK-26227 #close
Reported by Mark Michelson

Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9

The new endpoint option allows the PJSIP channel driver's fax_detect
endpoint option to timeout on a call after the specified number of
seconds into a call.  The new feature is disabled if the timeout is set
to zero.  The option is disabled by default.

ASTERISK-26214
Reported by: Richard Mudgett

Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d

Since 5th November 2014, the master branch of libSRTP changed the prefix of
several member names and is not compatible with the source code in Asterisk
anymore. Therefore instead, this change checks out the latest version of the
libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as
backend. This makes AES-GCM and AES-IN possible.

ASTERISK-22131 #close

Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6

If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.

ASTERISK-25471 #close

Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514

The patch removes updating all Endpoints' status on startup.
Instead, only non-qualified aors with static contact
and non-qualified non-expired contacts are retrieved from the realtime to
update the endpoint status to ONLINE.
The endpoint name was added to the contact object to simply find the endpoint
that created this contact.

The status of endpoints with qualified aors will be updated by 'qualify'
functions.

ASTERISK-26061 #close

Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df