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chan_sip deprecation.

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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
This commit is contained in:
Corey Farrell
2018-10-10 08:09:15 -04:00
parent e81d33e78f
commit 90a11c4ae7
10 changed files with 65 additions and 9 deletions
Download Patch File Download Diff File Expand all files Collapse all files

11
CHANGES
View File

@@ -8,6 +8,17 @@
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16 to Asterisk 17 --------------------
------------------------------------------------------------------------------
chan_sip
------------------
* The chan_sip module is now deprecated, users should migrate to the
replacement module chan_pjsip. See guides at the Asterisk Wiki:
https://wiki.asterisk.org/wiki/x/tAHOAQ
https://wiki.asterisk.org/wiki/x/hYCLAQ
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------
------------------------------------------------------------------------------

6
UPGRADE.txt
View File

@@ -27,6 +27,12 @@
New in 17.0.0:
chan_sip:
- The chan_sip module is now deprecated, users should migrate to the
replacement module chan_pjsip. See guides at the Asterisk Wiki:
https://wiki.asterisk.org/wiki/x/tAHOAQ
https://wiki.asterisk.org/wiki/x/hYCLAQ
func_callerid:
- The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been
removed.

41
channels/chan_sip.c
View File

@@ -176,7 +176,7 @@
/*** MODULEINFO
<use type="module">res_crypto</use>
<use type="module">res_http_websocket</use>
<support_level>extended</support_level>
<support_level>deprecated</support_level>
***/
/*! \page sip_session_timers SIP Session Timers in Asterisk Chan_sip
@@ -35321,6 +35321,37 @@ static const struct ast_sip_api_tech chan_sip_api_provider = {
.sipinfo_send = sipinfo_send,
};
static void deprecation_notice(void)
{
ast_log(LOG_WARNING, "chan_sip has no official maintainer and is deprecated. Migration to\n");
ast_log(LOG_WARNING, "chan_pjsip is recommended. See guides at the Asterisk Wiki:\n");
ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/x/tAHOAQ\n");
ast_log(LOG_WARNING, "https://wiki.asterisk.org/wiki/x/hYCLAQ\n");
}
/*! \brief Event callback which indicates we're fully booted */
static void startup_event_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message)
{
struct ast_json_payload *payload;
const char *type;
if (stasis_message_type(message) != ast_manager_get_generic_type()) {
return;
}
payload = stasis_message_data(message);
type = ast_json_string_get(ast_json_object_get(payload->json, "type"));
if (strcmp(type, "FullyBooted")) {
return;
}
deprecation_notice();
stasis_unsubscribe(sub);
}
static int unload_module(void);
/*!
@@ -35534,6 +35565,12 @@ static int load_module(void)
ast_websocket_add_protocol("sip", sip_websocket_callback);
}
if (ast_fully_booted) {
deprecation_notice();
} else {
stasis_subscribe_pool(ast_manager_get_topic(), startup_event_cb, NULL);
}
return AST_MODULE_LOAD_SUCCESS;
}
@@ -35751,7 +35788,7 @@ static int unload_module(void)
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Session Initiation Protocol (SIP)",
.support_level = AST_MODULE_SUPPORT_CORE,
.support_level = AST_MODULE_SUPPORT_DEPRECATED,
.load = load_module,
.unload = unload_module,
.reload = reload,

2
channels/sip/config_parser.c
View File

@@ -20,7 +20,7 @@
*/
/*** MODULEINFO
<support_level>extended</support_level>
<support_level>deprecated</support_level>
***/
#include "asterisk.h"

2
channels/sip/dialplan_functions.c
View File

@@ -20,7 +20,7 @@
*/
/*** MODULEINFO
<support_level>extended</support_level>
<support_level>deprecated</support_level>
***/
/*** DOCUMENTATION

2
channels/sip/reqresp_parser.c
View File

@@ -20,7 +20,7 @@
*/
/*** MODULEINFO
<support_level>extended</support_level>
<support_level>deprecated</support_level>
***/
#include "asterisk.h"

2
channels/sip/route.c
View File

@@ -20,7 +20,7 @@
*/
/*** MODULEINFO
<support_level>extended</support_level>
<support_level>deprecated</support_level>
***/
#include "asterisk.h"

2
channels/sip/security_events.c
View File

@@ -25,7 +25,7 @@
*/
/*** MODULEINFO
<support_level>extended</support_level>
<support_level>deprecated</support_level>
***/
#include "asterisk.h"

2
channels/sip/utils.c
View File

@@ -22,7 +22,7 @@
*/
/*** MODULEINFO
<support_level>extended</support_level>
<support_level>deprecated</support_level>
***/
#include "asterisk.h"

4
configs/samples/modules.conf.sample
View File

@@ -41,4 +41,6 @@ noload => chan_console.so
noload => res_hep.so
noload => res_hep_pjsip.so
noload => res_hep_rtcp.so
;
; Do not load chan_sip by default, it may conflict with res_pjsip.
noload => chan_sip.so