Given a sip.conf with the following content:

setvar FOO=1
setvar BAR=42

I want my generated pjsip.conf to containt the following set_vars

set_var FOO=1
set_var BAR=42

in the matching endpoint section.

Change-Id: I6c822401fda4133c3b44bf31e655b4eb939d4d26

The module 'res_pjsip_notify' inefficiently makes a lot of DB requests
on CLI completion on the endpoint.

For example if there are 10k endpoints the module makes 10k requests
of these 10k records.

Even if a part of the endpoint entered
the module makes the same 10k requests and then filtered them by itself.

This patch gathers endpoints container by prefix
and adds all gathered endpoints to completion at once.

ASTERISK-28137 #close

Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b

Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header.  This allows chan_pjsip to have
the same behavour as chan_sip

ASTERISK-28087 #close

Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95

Add 'Section:' headings and use '-' for bullet points.

Change-Id: I7e2be35601ac8fea53b90d926da564512b6716e4

Change-Id: Ia155ce2a53d61556aa4685524d1b48cfacfa3a8b

Default logging was not setup correctly when there was no logger.conf.
This resulted in many expected log messages not actually getting out to
the console.

Change-Id: I542e61c03b2f630ff5327f9de5641d776c6fa70c

The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates
from the called parties to the caller.

This patch also blocks updates in the other direction before call is
answered.

ASTERISK-27980

Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01

Change-Id: Id449d4435c38148b56ac4cfd61ae4d90ac66bb90

Adding the "label" attribute used for participant info correlation
was previously done in app_confbridge but it wasn't working
correctly because it didn't have knowledge about which video
streams belonged to which channel.  Only bridge_softmix has that
data so now it's set when the bridge topology is changed.

ASTERISK-28107

Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499

Change-Id: Ia1b2b386505b3102136dab02c45eaaf09f0f89c5

This change implements a few different generic things which were brought
on by Google Voice SIP.

1.  The concept of flow transports have been introduced.  These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target.  These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity).  When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.

2.  Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.

3.  P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module.  If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.

4.  Configurable outbound extension support has been added to the outbound
registration module.  When set the extension will be placed in the
Supported header.

5.  Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.

6.  Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.

All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.

ASTERISK-27971 #close

Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58

These language codes (tw, ge, mx, and cz) were deprecated in 1.6.2.

Change-Id: I18e4d2af2e83556fa91e39a7338030583ef05d50

Change-Id: I1a00ca22d59d6b6d2166aa56f0e9338a33e5ac60

These macros have been documented as legacy for a long time but are
still used in new code because they exist.  Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc

These macro's are also removed.  Only ao2_container_alloc remains due to
it's use in over 100 places.

Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a

__ast_mutex_logger used the variable `canlog` without accepting it as a
argument.  Replace with internal macro `log_mutex_error` which takes
canlog as the first arguement.  This will prevent confusion when working
with lock.c code, many of the function declare the canlog variable and
in some cases it previously appeared to be unused.

Change-Id: I83b372cb0654c5c18eadc512f65a57fa6c2e9853

* Update the post-answer documentation and example.  The Dial example was
incorrect and misleading for the post-answer subroutine useage.

* Fix note and warning paragraphs in option descriptions.  They don't show
up in the wiki.

Change-Id: I81019a1fd75d5b9151f76b52c38e2a90da682d14

PARKINGSLOT was deprecated in Asterisk 12 but the sample config was not
updated to reflect that.

Change-Id: I3e087c19d9ee587094fa5304102d8084a79c2b3c

Change-Id: I9882394617724a497df1d6f529a87965191be3ce

Most were in comments.  A couple were in warning messages.

Pointed out by Jonathan H on the Asterisk users mailing list.

Change-Id: I6286939dff5d0a27a2758140570106f1cb351855

* The dependency ensures that res_pjproject cannot be manually unloaded
before res_rtp_asterisk.
* The dependency allows startup loading errors to report that
res_rtp_asterisk depends upon res_pjproject.

Change-Id: Icf5e7581f4ddd6189929f6174c74dd951f887377

Change-Id: I29b9dbfa4bbfc49f21eba356858e38b1d3041824