The Location headers returned by:

 * /bridges/{bridgeId}/play
 * /bridges/{bridgeId}/record
 * /channels/{channelId}/play
 * /channels/{channelId}/record

Did not have the '/ari' prefix, and in the case of the 'play' resources, were
using 'playback' instead of 'playbacks.'

Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c

In 13.9.0, there was an issue where PJSIP contacts added to an AOR would
be deleted at seemingly random times.

One reason this was happening was because of an operation to retrieve
the contacts whose expiration time was less than or equal to the current
time. When retrieving existing contacts, the contact's expiration time
and the current time were converted from a string to a float, and those
two floats were compared.

On some systems, including mine, this conversion was horribly off. For
instance, I could regularly see the string "1463079214" get converted
into 1463079168.000000. When switching from using a float to using a
double, the conversion was as expected.

Why was the conversion to float off? My best guess is that the
conversion to float was attempting to store the entire value in the 23
bit significand of the IEEE-754 floating point number. In particular, if
you take only the 23 most significant bits of 1463079214, you get the
messed up 1463079168 that we were seeing in the conversion. It likely
was possible to get a more precise value by composing the number using
an exponent, but the conversion did not work that way. With a double,
you have a 52 bit significand, allowing the entire value to fit there,
and thereby allowing an accurate conversion.

ASTERISK-26007 #close
Reported by Greg Siemon

Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070

There was a newline missing from the end of the "no matching endpoint" notice.

Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181

There are two types of SIP URIs indicating a secure transport:
* sips:user@example.org
* sip:user@example.org;transport=tls

When using a sips URI, Asterisk checks incoming INVITEs and answers from
the other side for sips URIs, and rejects the packet if there are only
sip URIs. So Asterisk should only generate a sips Contact URI if the
other side supports it.

This patch makes Asterisk generate either a sip or sips Contact URI
depending on the format of the server URI.

If you want a sip URI, use:
server_uri=sip:example.org\;transport=tls

If you want a sips URI, use:
server_uri=sips:example.org

ASTERISK-25990 #close
Reported-by: Sebastian Damm

Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2

When running on a system that does not support or use AST_UNDEFINED_SANITIZER
or AST_LEAK_SANITIZER, the configure script would incorrectly set those
constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would
cause menuselect to error out, complaining that a blank value is not a
valid option. This patch corrects the issue by setting the value to 0 if
the options that those constants enable/disable is not found.

Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba

FollowMe with the option a records the name of the caller and plays it
to the callee. However it has failed to clean up that recorded file
as it tried to delete the file name without the '.sln' extension.

ASTERISK-26008 #close

Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

* No need to set language in a miniml configuration. 'en' will do just
  fine.
* It would be useful to have an example of setting it to a different
  language.
* Setting the documentation language explicitly is likewise not
  required. Setting it to a different value is not common. At least
  until there is a set of translated documentation.

Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

Don't suggest users to use debug level 5, which spews (usually
non-useful) debug information. Reduce the suggestion to (an
arbitrarily-selected) level 2.

Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

Note the default of remmed-out options. To clarify that those values are
not the defaults.

Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

A minimal configuration does not need to explicitly spell out the
directories. The built-in defaults will do just fine. In many cases
they are wrong.

Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

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

The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2.
SSL is not allowed.   So, even if you specify "sslv3" for a transport method,
it's silently ignored and one of the TLS protocols is used.  This was a new
behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that
we never caught.

Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default().
This tells pjproject to set the socket protocol to match the method.

ASTERISK-26004 #close

Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078

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