Log message:
asterisk21: reset PKGREVISION after update

Log message:
Update to Asterisk 21.3.1:  various bug fixes and minor improvements

## Change Log for Release asterisk-21.3.1

### Links:

 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - \ 
[GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9): \ 
res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests

### Commits By Author:

- ### George Joseph (1):
  - Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier \ 
transport ad..

## Change Log for Release asterisk-21.3.0

### Links:

 - [Full \ 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)

### Summary:

- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0

### User Notes:

- #### res_pjsip_logger: Preserve logging state on reloads.
  Issuing "pjsip reload" will no longer disable
  logging if it was previously enabled from the CLI.

- #### loader.c: Allow dependent modules to be unloaded recursively.
  In certain circumstances, modules with dependency relations
  can have their dependents automatically recursively unloaded and loaded
  again using the "module refresh" CLI command or the ModuleLoad AMI \ 
command.

- #### tcptls/iostream:  Add support for setting SNI on client TLS connections
  Secure websocket client connections now send SNI in
  the TLS client hello.

- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
  set identify_by=transport for the pjsip endpoint. Then
  use the existing 'match' option and the new 'transport' option of
  the identify.
  Fixes: #672

- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
  this new feature let users match endpoints based on the
  indound SIP requests' URI. To do so, add 'request_uri' to the
  endpoint's 'identify_by' option. The 'match_request_uri' option of
  the identify can be an exact match for the entire request uri, or a
  regular expression (between slashes). It's quite similar to the
  header identifer.
  Fixes: #599

- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
  the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.

- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
  When using the Originate AMI Action, we now can pass the PreDialGoSub
  parameter, instructing the asterisk to perform an subrouting at
  channel before call start. With this parameter an call initiated
  by AMI can request the channel to start the call automaticaly,
  adding a SIP header to using GoSUB, instructing to autoanswer
  the channel, and proceeding the outbuound extension executing.
  Exemple of an context to perform the previus indication:
  [addautoanswer]
  exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
  exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
  exten => _s,n,Return()

- #### manager.c: Add CLI command to kick AMI sessions.
  The "manager kick session" CLI command now
  allows kicking a specified AMI session.

- #### chan_dahdi: Allow specifying waitfordialtone per call.
  "waitfordialtone" may now be specified for DAHDI
  trunk channels on a per-call basis using the CHANNEL function.

- #### Upgrade bundled pjproject to 2.14.1
  Bundled pjproject has been upgraded to 2.14.1. For more
  information visit pjproject Github page: \ 
https://github.com/pjsip/pjproject/releases/tag/2.14.1

### Upgrade Notes:

- #### pbx_variables.c: Prevent SEGV due to stack overflow.
  The maximum amount of dialplan recursion
  using variable substitution (such as by using EVAL_EXTEN)
  is capped at 15.

Log message:
*: recursive bump for gnutls p11-kit option

(existing installations need the bl3.mk included, but it's now only
optionally included)

Log message:
comms/asterisk21: import asterisk-21.2.0

Asterisk is a complete PBX in software.  It provides all of the
features you would expect from a PBX and more. Asterisk does voice
over IP in three protocols, and can interoperate with almost all
standards-based telephony equipment using relatively inexpensive
hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and IAX.

This is an standard version.  It is secheduled to go to security
fixes only on November 18th, 2025, and EOL on November 18th, 2026.
See here for more information about Asterisk versions:
https://docs.asterisk.org/About-the-Project/Asterisk-Versions

Note that many things that have long been deprecated have now been
removed, such as chan_sip and app_macro.  See here for a complete
list:  http://docs.asterisk.org/Development/Asterisk-Module-Deprecations