Merge storage for the stats object and name string into the main
allocation for struct ast_taskprocessor.

Change-Id: I74fe9a7f357f0e6d63152f163cf5eef6428218e1

When Asterisk's taskprocessors get overloaded we need to reduce the work
load. res_pjsip currently ignores new SIP requests and relies on SIP
retransmissions in the hope that the overload condition will clear soon
enough to handle the retransmitted SIP request.
This change adds the following code after ast_taskprocessor_alert_get()
has returned TRUE:
1- identifies transport type. If non-udp then send a 503 response
2- if transport type is udp/udp6 then ignore, as before.

Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836

The use of a '|' in the "global/debug" synopsis documentation caused the
generated html table on the wiki to add an extra column that included the
text after the pipe.

This patch replaces the pipe with a comma.

ASTERISK-28150

Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719

The current round-robin method does not take the current taskprocessor
load into consideration when distributing requests.  Using the least-size
method the request goes to the taskprocessor that is servicing the least
number of active tasks at the current time.

Longer running tasks with the round-robin method can delay processing
tasks.

* Change the algorithm from round-robin to least-size for picking the
PJSIP taskprocessor from the default serializer pool.

Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd

As mentioned in the comment I've added in the code there is no
ability to unsubscribe all subscribers from a topic and explicitly
destroy it. This is not currently a problem as we have two types of
topics:

Long lived topics which exist for the lifetime of the system.
Ephemeral topics which feed a long lived topic.

In the case of the ephemeral topics there is no subscriber which does
not have its lifetime managed by the same entity that has created
the topic. This ensures that when the topic is being unreferenced the
subscribers are also unsubscribed and destroyed, allowing the topic
to ultimately be destroyed as well.

Change-Id: Ic5e244da7b16b1895ba1fc5ece481ebba5809c9a

When a call pickup is performed using and invite with replaces header
the ast_do_pickup method is attempted and a PICKUP stasis message is sent.

ASTERISK-28081 #close
Reported-by: Luit van Drongelen

Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a

Using the --quiet or -q option in conjonction with /dev/stdout as the output
file allow the output to be used as a valid configuration.

Given a script that generates a valid sip.conf I can pipe the output of that
script into `sip_to_pjsip.py -q /dev/stdin /dev/stdout`. This allow me to use
that piped command in my pjsip.conf using the `exec` command.

ASTERISK-28136

Change-Id: I7b0e2e90e2549f3f8e01dc96701f111b5874c88d

ASTERISK-28087

Change-Id: I69d48813ec514f5ef06c6de994cba52630e0a3b4

ASTERISK-28087

Change-Id: I046d018015427d0916fab571b5a4f5367476f729

This patch adds new options 'trust_connected_line' and 'send_connected_line'
to the endpoint.

The option 'trust_connected_line' is to control if connected line updates
are accepted from this endpoint.

The option 'send_connected_line' is to control if connected line updates
can be sent to this endpoint.

The default value is 'yes' for both options.

Change-Id: I16af967815efd904597ec2f033337e4333d097cd

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

This officially deprecates chan_sip in Asterisk 17+.  A warning is
printed upon startup or module load to tell users that they should
consider migrating.  chan_sip is still built by default but the default
modules.conf skips loading it at startup.

Very important to note we are not scheduling a time where chan_sip will
be removed.  The goal of this change is to accurately inform end users
of the current state of chan_sip and encourage movement to the fully
supported chan_pjsip.

Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93

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