From the issue reporter:
"res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of
the timestamp, the source address, the source port, a server UUID that is
calculated at startup, and the authentication realm.

Rather than caching nonces that we create, we instead attempt to re-calculate
the nonce when receiving an incoming request with authentication. We then
compare the re-calculated nonce to the incoming nonce, and if they don't match,
then authentication has failed early.

The problem is that it is possible, especially when using TCP, to receive two
requests from the same endpoint but have differing source ports for those
requests. Asterisk itself commonly will use different source ports for
outbound TCP requests."

This patch removes the source port dependency when building the nonce.

ASTERISK-25978 #close

Change-Id: I871b5f4adce102df1c4988066283095ec509dffe

This migrates res_pjsip_pubsub over to using the newly
introduce common datastores management API instead of using
its own implementations for both subscriptions and
publications.

As well the extension state data now provides a generic
datastores container instead of a subscription. This allows
the dialog-info+xml body generator to work for both
subscriptions and publications.

ASTERISK-25999 #close

Change-Id: I773f9e4f35092da0f653566736a8647e8cfebef1

This change introduces a common container based datastores
management API. This has been done in a few places across
the tree but this consolidates all of the logic into one
place in a generic fashion.

ASTERISK-25999

Change-Id: I72eb15941dcdbc2a37bb00a33ce00f8755bd336a

This patch allows for having app_confbridge register the name of the
conference as an extension into a specific context, similar to
regcontext for chan_sip.  This variant is not quite as involved as the
one in chan_sip and doesn't allow for multiple contexts or custom
extensions, you can only specify the context and the conference name
will always be used as the extension to register.

ASTERISK-25989 #close

Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f

The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set.
The python bindings are now built only if TEST_FRAMEWORK is set and a
python development package is installed.

libresample was also disabled.

ASTERISK-25993 #close
Reported-by: Joshua Colp

Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03

The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_*
and res_pjsip_registrar modules should load ASAP
to avoid "No matching endpoint found" for legitimate endpoint.

ASTERISK-25994

Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b

ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail.
ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail.
These additions should be also in stasis_endpoints
to include in command "manager show event ContactStatus"

Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a

When res_pjsip_outbound_publish unloads it has to wait for all current
publishing objects to get done. However if the wait condition times out
then it does not fail the unload. This sometimes results in an infinite
loop check while unloading. This patch now fails the unload operation if
the condition times out.

Change-Id: Id57b8cbed9d61222690fcba1e4f18e259df4c7ec

There were a few spots where the client object's reference was being leaked in
sip_outbound_publish_callback. This patch cleans up those leaks.

Change-Id: I485d0bc9335090f373026f77c548042e258461df

It was possible for the explicit publish destroy function to be called without
the pjsip client ever being initialized. This fix checks to make sure there is
a client to destroy before attempting.

Change-Id: I8eea1bfa3bd472149bfc255310be2a6248688f5c

It is possible for the nativeformats of a channel to change
throughout its lifetime. As a result a user of it needs to either
ensure the channel is locked when accessing the formats or keep
a reference to the nativeformats themselves.

This change fixes the file playback support so it keeps a
reference to the nativeformats when accessing things.

ASTERISK-25998 #close

Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915

This patch modified pjsip_options to retrieve only
permament contacts for aor if the qualify_frequency is > 0
and persisted contacts if the qualify_frequency is > 0.

This patch also fixed a bug in res_sorcery_astdb.
res_sorcery_astdb doesn't save object data retrived from astdb.

ASTERISK-25826

Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05

The channel and peer V.21 sessions are created on the same channel now.
The peer V.21 session should be created only on peer channel
when one of channel can handle T.38.

Also this patch enable debug for T.38 gateway session
if global fax debug enabled.

ASTERISK-25982

Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e

ASTERISK-25931

Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549

ASTERISK-25956 #close

Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38

The app_fax set FAXMODE variable, but res_fax missing this feature.
This patch add FAXMODE variable which is set to either "audio" or "T38".

ASTERISK-25980

Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b

When option 'o' was not set, ChanSpy created its audiohook with the flag
AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one
direction only.

ASTERISK-25866 #close

Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0

With the old SIP module AMI sends PeerStatus event on every
successfully REGISTER requests, ie, on start registration,
update registration and stop registration.

With PJSIP AMI sends ContactStatus only when status is changed.
Regarding registration:
on start registration - Created
on stop registration - Removed
but on update registration nothing

This patch added contact.updated event.

ASTERISK-25904

Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f

For all OSes:
* Disabled third-party codecs in pjproject and added
  '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the
  configure options since we don't use the pjsip codec capability.

FreeBSD:
* Added FreeBSD support to install_prereq.
* Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make".
* Added __progname and environ to asterisk.exports.in.
* Reverted the use of ldconfig to create shared library symlinks to ln.
* Only enable epoll in pjproject if `uname -s` is Linux.
* Added a patch to pjproject to take the name of the 'make' command from
  an environment variable if supplied.  This is needed for the python bindings.
  (merged by Teluu into pjproject trunk 5/3/2016)
FreeBSD support isn't complete.  Still some general issues regarding
make/gmake having nothing to do with pjproject.  With some handholding it DOES
build successfully.

CentOS:
Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH.
CentOS 6/7 32/64 build and run the pjsip testsuite successfully.

Ubuntu:
No changes required.
Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully.

Debian:
No changes required.
Debian 6/7/8 32/64 build and run the pjsip testsuite successfully.

There will utimately be a follow-up patch to create an install_prereq for
the testsuite as I've discovered a few missing requirements.

ASTERISK-25968 #close

Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c

Voicemail email addresses can be corrupt or voicemail
emails can end up being sent to the wrong email address if asterisk is
reading voicemail.conf during a reload and processing an email at the
same time. This patch always copies the struct that would otherwise only
be copied once.

ASTERISK-24463 #close
Reported by: John Campbell
Tested by: Etienne Lessard
Tested by: Andrew Nagy
Change-Id: I3a0643813116da84e2617291903d0d489b7425fb