Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998

The downgrade function was missing "_v2" at the end of the
alter column type.

Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e

In 45744fc5, I mistakenly broke SDP media address rewriting by
misinterpreting which address was checked in the localnet comparison.

Instead of checking the remote peer address to decide whether we need
media address rewriting, we check our local media address: if it's
local, then we rewrite. This feels awkward, but works and even made
directmedia work properly if you set local_net. (For the record: for
local peers, the SDP media rewrite code is not called, so the
comparison does no harm there.)

ASTERISK-27248 #close

Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f

Change-Id: I43f25976aa3069793ddbe0086833965a6fb0a518

MS-SQL has no native Enum-type support and therefore
needs to work with constraints.
Since these constraints need unique names the suggested approach
referenced in the following alembic documentation has been applied:
http://bit.ly/2x9r8pb

ASTERISK-27255 #close

Change-Id: I8b579750dae0c549f1103ee50172644afb9b2f95

Fixes ${CDR(...,u)} when used in cdr_custom.conf

ASTERISK-27165 #close

Change-Id: Ia4e0b6ba93e03d27886354c279737790e2cd6a83

* WaitForSilence completes successfully if it receives no media in the
  specified timeout, but when acting as WaitForNoise that logic needs
  to be reversed.

* Use standard argument parsing macros and add some error checking for
  invalid values.

* The documentation indicated that the first argument to both
  WaitForSilence and WaitForNoise was required when it was not. Update
  the documentation to reflect that.

* Wrap up some behavior in structs to avoid boolean checks all over the
  place.

ASTERISK-24066 #close
Reported by: M vd S

Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9

In handle_request_invite, when processing a pickup, a call
is made to get_sip_pvt_from_replaces to locate the pvt for
the subscription. The pvt is assumed to be valid when zero
is returned indicating no error, and is dereferenced which
can cause a crash if it was not found.

This change checks the not found case and returns -1 which
allows the calling code to fail appropriately.

ASTERISK-27217 #close
Reported-by: Bryan Walters

Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612

Change-Id: Ia0edb7dc0dbbb879c079ff7000f1b722d86ce7dc

If an error occurs during a bridge impart it's possible that
the "bridge_after" callback might try to run before
control_swap_channel_in_bridge has been signalled to continue.
Since control_swap_channel_in_bridge is holding the control lock
and the callback needs it, a deadlock will occur.

* control_swap_channel_in_bridge now only holds the control
  lock while it's actually modifying the control structure and
  releases it while the bridge impart is running.
* bridge_after_cb is now tolerant of impart failures.

Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3

If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE
to both parties to set up media path directly between the endpoints.
In this reINVITE msg SDP origin line (o=) contains IP address of endpoint
instead of IP of asterisk. This behavior violates RFC3264, sec 8:
"When issuing an offer that modifies the session,
the "o=" line of the new SDP MUST be identical to that in the
previous SDP, except that the version in the origin field MUST
increment by one from the previous SDP."
This patch assures IP address of Asterisk is always sent in
SDP origin line.

ASTERISK-17540
Reported by:  saghul

Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e

Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db

Added include for postgresql ENUM type and
redefined values in the same way as in the
other migration scripts.

ASTERISK-27254 #close

Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa

Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro
SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with
previous versions of libSRTP, this change allows libSRTP 2.1.x as well.

ASTERISK-27253 #close

Change-Id: I2e6eb3c3bc844fee8a624060a2eb6f182dc70315

When rtp_keepalive is on for a PJSIP endpoint dialing to another
Asterisk instance also using PJSIP, Asterisk will continue to print
warning messages about not being able to send frames of a certain
type. This suppresses that warning message.

Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67

Some formats are able to handle short reads while others are not, so
restore the previous behavior for the format modules so that we don't
have spurious errors when playing back files.

ASTERISK-27232 #close
Reported by: Jens T.

Change-Id: Iab7f52b25a394f277566c8a2a4b15a692280a300

In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was
confusion about whether the transport_state->localnet ACL has ALLOW or
DENY semantics.

For the record: the localnet has DENY semantics, meaning that "not in
the list" means ALLOW, and the local nets are in the list.

Therefore, checks like this look wrong, but are right:

    /* See if where we are sending this request is local or not, and if
       not that we can get a Contact URI to modify */
    if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
        ast_debug(5, "Request is being sent to local address, "
                     "skipping NAT manipulation\n");

(In the list == localnet == DENY == skip NAT manipulation.)

And conversely, other checks that looked right, were wrong.

This change adds two macro's to reduce the confusion and uses those
instead:

    ast_sip_transport_is_nonlocal(transport_state, addr)
    ast_sip_transport_is_local(transport_state, addr)

ASTERISK-27248 #close

Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934

Stream names within Asterisk can have meaning so when an externally
initiated renegotiation occurs we need to preserve the name of
the stream if it already exists.

Change-Id: I29f50d0cc7f3238287d6d647777e76e1bdf8c596

Change-Id: I31eee8be30c6b0fc3dadb31111dd47742da8892d

t38_reinvite_response_cb can get called by res_pjsip_session's
session_inv_on_tsx_state_changed in situations where session->channel
is NULL.  If it is, the ast_log warning segfaults because it tries
to get the channel name from a NULL channel.

* Check session->channel and print "unknown channel" when it's NULL.

ASTERISK-27236
Reported by: Ross Beer

Change-Id: I4326e288d36327f6c79ab52226d54905cdc87dc7

ASTERISK-27225 #close
Reported by: Richard Kenner

Change-Id: I097b81734ef730f8603c0b972909d212a3a5cf89

ASTERISK-27177 #close
Reported by: Tzafrir Cohen

Change-Id: I40311c404edb2302a7543ad5ca7a06b2a38f2d97