Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function
instead of correct 'dtmf_features'

ASTERISK-27377 #close

Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930

ASTERISK-27181

Change-Id: Ic4468b49860bd7f67e922baf4c9e96828c184d17

Earlier versions of the codec_opus samples_count callback can return
negative error values on undecodable frames.  This resulted in a divide by
zero exception.

* Added a defensive check in ast_codec_samples_count() for a "negative"
samples count return value.  Log the event and set the count to zero.

ASTERISK-27194

Change-Id: Icf69350307ecbbc80a3d74de46af9bd80ea17819

When the identify_by option on an endpoint is set to ip it will
only be identified using the res_pjsip_endpoint_identifier_ip module.
This ensures that it is not mistakenly matched using the username of
the From header. To ensure behavior has not changed the default has
been changed to "username,ip" for the identify_by option.

ASTERISK-27206

Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd

Currently ast_http_send barricades a portion of the content that
needs to be sent in order to establish a connection for things
like the ARI client. The conditional and contents have been changed
to ensure that everything that needs to be sent, will be sent.

ASTERISK-27372

Change-Id: I8816d2d8f80f4fefc6dcae4b5fdfc97f1e46496d

This makes the 'bt' parameter unconditional for ast_store_lock_info and
ast_remove_lock_info.  The 'bt' parameter is unused when HAVE_BKTR is
undefined.

Change-Id: Ieced0e920928b735a39c3b5952b806c473d67453

When chan_sip receives a SUBSCRIBE request with no "Expires" header it
processes the request as an unsubscribe.  This is incorrect, per RFC3264
when the "Expires" header is missing a default expiry should be used.

ASTERISK-18140

Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5

ASTERISK-23556
Reported by: Marcello Ceschia

Change-Id: Ic27e88e0336a0d83877dc857938659dc5560b93c

A few places in hashtab use free instead of ast_free, remove declaration
of ASTMM_LIBC from hashtab.c as it's no longer needed.

Change-Id: I2ff089bad71640c03c3ce97f1b00fc962ef79427

ASTERISK-27359

Change-Id: Ib01fb6c01f9bb87129374a51cb9318c474147517

Change-Id: Ibb3e47f27a395d74d8c5263db015b05434f5969b

On second run the config_hook test was unexpectedly failing to load
test_config.conf because it was still unmodified since the last load.
This is fixed by not passing CONFIG_FLAG_FILEUNCHANGED for the initial
loads, only using it when we are tested that a reload of unmodified
files do not initiate the hook.

ASTERISK-25960

Change-Id: Ifd679509a23ed163e5cc647490bf7df4ae3cd856

create_outgoing_sdp_stream was setting "addr_type = STR_IP6" only
when an ipv6 media_address was specified on the endpoint.  If
rtp_ipv6 was set and ast_sip_get_host_ip_string returned an ipv6
address, we were leaving the addr_type set at the default of
STR_IP4.  This caused the address type to be set incorrectly on the
"o" and "c" SDP attributes even though the address was set
correctly.  Some clients don't like the mismatch.

 * Removed the test for endpoint/media_address and now check all
   addresses for ipv6.

ASTERISK-27198
Reported by: Martin Cisárik

Change-Id: I5214fc31b728117842243807e7927a319cf77592

Change-Id: Ib0bc95fd0ec288c78c313823254d7a84ebfc4429

Users of the API that res_xmpp provides expect that a
filter be available on the client at all times. When
OAuth authentication support was added this requirement
was not maintained.

This change merely moves the OAuth authentication to
after the filter is created, ensuring users of res_xmpp
can add things to the filter as needed.

ASTERISK-27346

Change-Id: I4ac474afe220e833288ff574e32e2b9a23394886

This matches the behavior of the other SIP channel driver, chan_pjsip.

ASTERISK-27365

Change-Id: I8f23a51290a58b75816da2999ed1965441dfc5d6

Update patches included in bundled PJPROJECT for the new version.

ASTERISK-27355

Change-Id: I9ac5dbbffaadca25ad24fac8b9ab615e5ace6083

Prevent unload of the module as certain pjsip initialization functions
cannot be reversed.  This required a reorder of the module_load so that
the non-reversable pjsip functions are not called until all potential
errors have been ruled out.

ASTERISK-24483

Change-Id: Iee900f20bdd6ee1bfe23efdec0d87765eadce8a7

Prevent unload of the module as certain pjsip initialization functions
cannot be reversed.

ASTERISK-24483

Change-Id: I94597ec8b8491f5af9c57bf66dbc3b078fe2d49d

When sip.conf contains 'sipdebug=yes' it is impossible to disable it
using CLI 'sip set debug off'.  This corrects the output of that CLI
command to instruct the user to turn sipdebug off in the configuration
file.

ASTERISK-23462 #close

Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318

* Mark the module deprecated.
* Disable the module by default.
* Produce a warning the first time a macro is used.
* Note deprecation related options in app_dial and app_queue.

ASTERISK-27350

Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc

Beside allowing AES-GCM again, this adds AES-192 again.

ASTERISK-27356

Change-Id: Ia97a435faf26300335d9552fa676b5d17e5f7233