Modules commonly used the pj_gethostip function for retrieving the
IP address of the host. This function does not cache the result and may
result in a DNS lookup occurring, or additional work. If the DNS
server is unreachable or network issues arise this can cause the
pj_gethostip function to block for a period of time.

This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string
function which does the same thing but caches the host IP address at
module load time. This results in no additional work being done each
time the local host IP address is needed.

ASTERISK-25342 #close

Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e

When executing an action in a bridge it is possible for the
channel to be hung up without the bridge becoming aware of it.
This is most easily reproducible by hanging up when the bridge
is streaming DTMF due to a feature timeout. This change makes
it so after action execution the channel is checked to determine
if it has been hung up and if it has it is kicked from the bridge.

ASTERISK-25341 #close

Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062

When recreating a subscription it is possible for a freed sub_tree
to be referenced when the initial NOTIFY fails to be created.

Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788

When an endpoint is backed by a non-static conf file backend (such as
the AstDB or Realtime), the 'auth' object may be returned as being an
empty string. Currently, res_pjsip will interpret that as being a valid
auth object, and will attempt to authenticate inbound requests. This
isn't desired; is an auth value is empty (which the name of an auth
object cannot be), we should instead interpret that as being an invalid
auth object and skip it.

ASTERISK-25339 #close

Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7

Change-Id: I18b7d75187548a9ed55b4f258d21aaaf29d08874

ASTERISK-25166
Reported by: Kevin Harwell

ASTERISK-17410
Reported by: Boris Fox

Change-Id: I7f04d5c8bee1126fee5fe6afbc39e45104469f4e

ASTERISK-25166
Reported by: Kevin Harwell

ASTERISK-17410
Reported by: Boris Fox

Change-Id: I97ecebc1ab9b5654fb918bf1f4c98c956b852369

* Make ast_rtp_codecs_payload_code() get the current mapping or create a
rx payload type mapping.

ASTERISK-25166
Reported by: Kevin Harwell

ASTERISK-17410
Reported by: Boris Fox

Change-Id: Ia4b2d45877a8f004f6ce3840e3d8afe533384e56

ASTERISK-25166
Reported by: Kevin Harwell

ASTERISK-17410
Reported by: Boris Fox

Change-Id: I34e23bf5b084c8570f9c3e6ccd19b95fe85af239

There are numerous problems with the current implementation of the RTP
payload type mapping in Asterisk.  It uses only one mapping structure to
associate payload types to codecs.  The single mapping is overkill if all
of the payload type values are well known values.  Dynamic payload type
mappings do not work as well with the single mapping because RFC3264
allows each side of the link to negotiate different dynamic mappings for
what they want to receive.  Not only could you have the same codec mapped
for sending and receiving on different payload types you could wind up
with the same payload type mapped to different codecs for each direction.

1) An independent payload type mapping is needed for sending and
receiving.

2) The receive mapping needs to keep track of previous mappings because of
the slack to when negotiation happens and current packets in flight using
the old mapping arrive.

3) The transmit mapping only needs to keep track of the current negotiated
values since we are sending the packets and know when the switchover takes
place.

* Needed to create ast_rtp_codecs_payload_code_tx() and make some callers
use the new function because ast_rtp_codecs_payload_code() was used for
mappings in both directions.

* Needed to create ast_rtp_codecs_payloads_xover() for cases where we need
to pass preferred codec mappings to the peer channel for early media
bridging or when we need to prefer the offered mapping that RFC3264 says
we SHOULD use.

* ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are
the only new public functions created.  All the others were only used for
the tx or rx mapping direction so the function doxygen now reflects which
direction the function operates.

* chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing
that makes no sense when processing an incoming SDP.  We would be wiping
out any mappings that we set for the possible outgoing SDP we sent
earlier.

ASTERISK-25166
Reported by: Kevin Harwell

ASTERISK-17410
Reported by: Boris Fox

Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac

This is a type mismatch fix of the debugging commit
c63316ee made to find out why
a testsuite test was failing only on one of the continuous
integration build agents.

Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75

Asterisk needs the sqlite 3 library, which is package
sqlite-devel in CentOS. By adding this package to the
script, a problem with configure failing is resolved.

ASTERISK-25331 #close
Reported by: Kevin Harwell

Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec

ASTERISK-25308 #close
Reported by: Joshua Colp

Change-Id: I592785bf70ff4b63d00e535b482f40da8e82a082

Change-Id: I228df6adecd4cb450d03e09e9a38c86bb566e811

* Remove extraneous unlock on off-nominal path.
* Add missing HTTP error reply.

Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b

Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf

Setting the 'paused' and 'ringinuse' options on a queue member using the
dialplan function QUEUE_MEMBER did not behave the same way as the
equivalent dialplan applications or AMI actions.

* Made queue_function_mem_write() call the set_member_paused() and
set_member_value() for the 'paused' and 'ringinuse' options respectively.
A beneficial side effect is that the queue name is now optional and sets
the value in all queues the interface is a member.

* Update QUEUE_MEMBER XML documentation.

* Fix error checking in QUEUE_MEMBER() write.

ASTERISK-25215 #close
Reported by: Lorne Gaetz

Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb

* Extract set_queue_member_pause() from set_member_paused() for simpler
and more consistent code.

* Extract set_queue_member_ringinuse() from
set_member_ringinuse_help_members() for simpler code.

Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306

Change-Id: I7294e13d27875851c2f4ef6818adba507509d224

When allocating a sorcery object, fail if the
id value was not allocated.

ASTERISK-25323
Reported by: Scott Griepentrog

Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e

When sending an RTP keepalive, we need to be sure we're not dealing with
a NULL RTP instance. There had been a NULL check, but the commit that
added the rtp_timeout and rtp_hold_timeout options removed the NULL
check.

Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64

Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c

The built frame format in audiohook_read_frame_both() is now set to a
signed linear format before the rx and tx frames are duplicated instead of
only for the mixed audio frame duplication.

ASTERISK-25322 #close
Reported by Sean Pimental

Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538

In chan_sip, after handling an incoming invite a security event is raised
describing authorization (success, failure, etc...). However, it was doing
a lookup of the peer by extension. This is fine for register messages, but
in the case of an invite it may search and find the wrong peer, or a non
existent one (for instance, in the case of call pickup). Also, if the peers
are configured through realtime this may cause an unnecessary database lookup
when caching is enabled.

This patch makes it so that sip_report_security_event searches by IP address
when looking for a peer instead of by extension after an invite is processed.

ASTERISK-25320 #close

Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4

Due to the use of ast_websocket_close in session termination it is
possible for the underlying socket to already be closed when the
session is terminated. This occurs when the close frame is attempted
to be written out but fails.

Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b