After configuring Asterisk with '--with-pjproject-bundled' the configure/build
process attempts to download pjproject from its download site. Currently, a
timeout of 10 seconds is used that will stop the download process if pjproject
has not been fully downloaded in that time. For some systems this was not enough
time and the process was timing out too early.

This patch raises the download timeout value to '60'. Also, this patch fixes
another bug where the DOWNLOAD_TIMEOUT variable was not being properly exported
due to a naming error. DOWNLOAD_MAX_TIMEOUT is now properly renamed to
DOWNLOAD_TIMEOUT.

ASTERISK-26814 #close

Change-Id: Ia56e4e8a3d39db76bc8a1852b2cf07ec10b39842

Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched
destinations) as currently provided by chan_sip is missing from res_pjsip.
This patch adds a new endpoint attribute (allow_overlap) [defaults to yes]
which when set to yes enables 484 responses to partial destination
matches rather than the current 404.

ASTERISK-26864

Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6

Rather than hard-coding UDP, allow consumers of the HEP API to specify
which protocol is in use. Update the PJSIP provider to pass in the
current protocol type.

ASTERISK-26850 #close

Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978

This reverts commit 163e9e53.

Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b

We aren't validating that the URI we just parsed is a SIP/SIPS one before
trying to access the user, host, and port members of a possibly uninitialized
structure.

Also update the MessageSend documentation to indicate what 'from' formats are
accepted.

ASTERISK-26484 #close
Reported by: Vinod Dharashive

Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30

ASTERISK-26776 #close

Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2

Some codecs - codec_speex specifically - take voice frames and return
other types of frames, like CNG. If we subsequently treat those as
voice frames, we'll run into trouble when destroying the frame because
of the requirement that each voice frame have an associated format.

ASTERISK-26880 #close
Reported by: Kirsty Tyerman

Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c

POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.

Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.

Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48

We are currently passing in the capacity of the read buffer instead of the
number of bytes that we actually read off the wire.

Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36

Queue member will get stuck in pending_members if queue calls a device
that is different from the one observed for state changes.

This patch removes members from pending_members as a result of channel stasis
events such as blind or attended transfers and hangup.

ASTERISK-26862 #close

Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727

* Added CHANNEL(callid) to retrieve the call identifier log tag associated
with the channel.  Dialplan now has access to the call log search key
associated with the channel so it can be saved in case there is a problem
with the call.

ASTERISK-26878

Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f

The queue_stasis_data structure contains various mutable fields that require
appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and
'caller_uniqueid' fields need to be locked when read from or written to.

Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088

ASTERISK-26846 #close

Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639

Thanks to Chris Howard for pointing this out on the wiki.

Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705

struct ast_rtcp does not define the dtls member if SRTP is not enabled.

ASTERISK-26732

Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e

We were inadvertenly referencing the cos_video option to determine if we
should set the tos_audio and cos_audio value on the RTP instance.

Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0

If local_net is not defined on a transport, transport_state->localnet
will be NULL. ast_apply_ha will, be default, return AST_SENSE_ALLOW in
this case, causing the external_media_address, if set, to be skipped.

This patch causes us to only check if we are sending within a network if
local_net is defined.

ASTERISK-26879 #close

Change-Id: Ib661c31a954cabc9c99f1f25c9c9a5c5b82cbbfb

Currently a wildcard address is used for the local RTP socket, which
will not always result in the same address as used by the SIP socket
(e.g. if explicit transport addresses are configured).
Use the transport's host address when binding new local RTP sockets if
available.

ASTERISK-26851

Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a

A new transport parameter 'symmetric_transport' has been added.

When a request from a dynamic contact comes in on a transport with
this option set to 'yes', the transport name will be saved and used
for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE.
It's saved as a contact uri parameter named 'x-ast-txp' and will
display with the contact uri in CLI, AMI, and ARI output.  On the
outgoing request, if a transport wasn't explicitly set on the
endpoint AND the request URI is not a hostname, the saved transport
will be used and the 'x-ast-txp' parameter stripped from the
outgoing packet.

* config_transport was modified to accept and store the new parameter.

* config_transport/transport_apply was updated to store the transport
  name in the pjsip_transport->info field using the pjsip_transport->pool
  on UDP transports.

* A 'multihomed_on_rx_message' function was added to
  pjsip_message_ip_updater that, for incoming requests, retrieves the
  transport name from pjsip_transport->info and retrieves the transport.
  If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter
  containing the transport name is added to the incoming Contact header.

* An 'ast_sip_get_transport_name' function was added to res_pjsip.
  It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a
  transport name if endpoint->transport is set or if there's an
  'x-ast-txp' parameter on the uri and the uri host is an ipv4 or
  ipv6 address.  Otherwise it returns NULL.

* An 'ast_sip_dlg_set_transport' function was added to res_pjsip
  which takes an ast_sip_endpoint, a pjsip_dialog, and an optional
  pjsip_tpselector.  It calls ast_sip_get_transport_name() and if
  a non-NULL is returned, sets the selector and sets the transport
  on the dialog.  If a selector was passed in, it's updated.

* res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas
  were modified to call ast_sip_dlg_set_transport() instead of their
  original logic.

* res_pjsip/create_out_of_dialog_request was modified to call
  ast_sip_get_transport_name() and pjsip_tx_data_set_transport()
  instead of its original logic.

* Existing transport logic was removed from endpt_send_request
  since that can only be called after a create_out_of_dialog_request.

* res_pjsip/ast_sip_create_rdata was converted to a wrapper around
  a new 'ast_sip_create_rdata_with_contact' function which allows
  a contact_uri to be specified in addition to the existing
  parameters.  (See below)

* res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated
  since all it did was transport selection and that is now done in
  ast_sip_create_dialog_uac and ast_sip_create_dialog_uas.

* 'contact_uri' was added to subscription_persistence.  This was
  necessary because although the parsed rdata contact header has the
  x-ast-txp parameter added (if appropriate),
  subscription_persistence_update stores the raw packet which
  doesn't have it.  subscription_persistence_recreate was then
  updated to call ast_sip_create_rdata_with_contact with the
  persisted contact_uri so the recreated subscription has the
  correct transport info to send the NOTIFYs.

* res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since
  all it did was transport selection and that is now done in
  ast_sip_create_dialog_uac.

* pjsip_message_ip_updater/multihomed_on_tx_message was updated
  to remove all traces of the x-ast-txp parameter from the
  outgoing headers.

NOTE:  This change does NOT modify the behavior of permanent
contacts specified on an aor.  To do so would require that the
permanent contact's contact uri be updated with the x-ast-txp
parameter and the aor sorcery object updated.  If we need to
persue this, we need to think about cloning permanent contacts into
the same store as the dynamic ones on an aor load so they can be
updated without disturbing the originally configured value.

You CAN add the x-ast-txp parameter to a permanent contact's uri
but it would be much simpler to just set endpoint->transport.

Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f

This change removes an assumption that when DTLS is stopped
an RTCP session will be present on the RTP session. This is not
always the case.

ASTERISK-26732

Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611