[svn-commits] kmoore: branch kmoore/channel_event_refactor r392857 - in /team/kmoore/channe...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jun 25 13:37:07 CDT 2013
Author: kmoore
Date: Tue Jun 25 13:37:04 2013
New Revision: 392857
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=392857
Log:
Rip out ChannelUpdate events and exclusively supporting code
Modified:
team/kmoore/channel_event_refactor/channels/chan_gtalk.c
team/kmoore/channel_event_refactor/channels/chan_iax2.c
team/kmoore/channel_event_refactor/channels/chan_sip.c
team/kmoore/channel_event_refactor/channels/sip/include/sip.h
team/kmoore/channel_event_refactor/configs/sip.conf.sample
Modified: team/kmoore/channel_event_refactor/channels/chan_gtalk.c
URL: http://svnview.digium.com/svn/asterisk/team/kmoore/channel_event_refactor/channels/chan_gtalk.c?view=diff&rev=392857&r1=392856&r2=392857
==============================================================================
--- team/kmoore/channel_event_refactor/channels/chan_gtalk.c (original)
+++ team/kmoore/channel_event_refactor/channels/chan_gtalk.c Tue Jun 25 13:37:04 2013
@@ -46,23 +46,6 @@
<use type="external">openssl</use>
<support_level>deprecated</support_level>
<replacement>chan_motif</replacement>
- ***/
-
-/*** DOCUMENTATION
- <managerEvent language="en_US" name="ChannelUpdate">
- <managerEventInstance class="EVENT_FLAG_SYSTEM">
- <synopsis>Raised when a GTalk SID is established for a call.</synopsis>
- <syntax>
- <xi:include xpointer="xpointer(/docs/managerEvent[@name='Newchannel']/managerEventInstance/syntax/parameter)" />
- <parameter name="Gtalk-SID">
- <para>The Gtalk session identifier.</para>
- </parameter>
- <parameter name="Channeltype">
- <para>The type of the channel (always <literal>Gtalk</literal>).</para>
- </parameter>
- </syntax>
- </managerEventInstance>
- </managerEvent>
***/
#include "asterisk.h"
@@ -273,11 +256,6 @@
static struct gtalk_container gtalk_list;
-static struct ast_manager_event_blob *channel_update_to_ami(struct stasis_message *msg);
-STASIS_MESSAGE_TYPE_DEFN_LOCAL(channel_update_type,
- .to_ami = channel_update_to_ami,
- );
-
static void gtalk_member_destroy(struct gtalk *obj)
{
obj->cap = ast_format_cap_destroy(obj->cap);
@@ -561,40 +539,6 @@
return IKS_FILTER_EAT;
}
-static struct ast_manager_event_blob *channel_update_to_ami(struct stasis_message *msg)
-{
- RAII_VAR(struct ast_str *, channel_string, NULL, ast_free);
- struct ast_channel_blob *obj = stasis_message_data(msg);
- const char *sid = ast_json_string_get(ast_json_object_get(obj->blob, "gtalk_sid"));
-
- if (obj->snapshot) {
- channel_string = ast_manager_build_channel_state_string(obj->snapshot);
- if (!channel_string) {
- return NULL;
- }
- }
-
- return ast_manager_event_blob_create(EVENT_FLAG_SYSTEM, "ChannelUpdate",
- "%s"
- "Channeltype: Gtalk\r\n"
- "Gtalk-SID: %s\r\n",
- S_COR(obj->snapshot, ast_str_buffer(channel_string), ""), sid);
-}
-
-static void send_channel_update(struct ast_channel *chan, const char *sid)
-{
- RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
-
- ast_assert(sid != NULL);
-
- blob = ast_json_pack("{s: s}", "gtalk_sid", sid);
- if (!blob) {
- return;
- }
-
- ast_channel_publish_blob(chan, channel_update_type(), blob);
-}
-
static int gtalk_answer(struct ast_channel *ast)
{
struct gtalk_pvt *p = ast_channel_tech_pvt(ast);
@@ -603,7 +547,6 @@
ast_debug(1, "Answer!\n");
ast_mutex_lock(&p->lock);
gtalk_invite(p, p->them, p->us,p->sid, 0);
- send_channel_update(ast, p->sid);
ast_mutex_unlock(&p->lock);
return res;
}
@@ -2376,10 +2319,6 @@
char *jabber_loaded = ast_module_helper("", "res_jabber.so", 0, 0, 0, 0);
struct ast_format tmpfmt;
- if (STASIS_MESSAGE_TYPE_INIT(channel_update_type)) {
- return AST_MODULE_LOAD_FAILURE;
- }
-
if (!(gtalk_tech.capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_DECLINE;
}
@@ -2477,7 +2416,6 @@
ASTOBJ_CONTAINER_DESTROY(>alk_list);
global_capability = ast_format_cap_destroy(global_capability);
gtalk_tech.capabilities = ast_format_cap_destroy(gtalk_tech.capabilities);
- STASIS_MESSAGE_TYPE_CLEANUP(channel_update_type);
return 0;
}
Modified: team/kmoore/channel_event_refactor/channels/chan_iax2.c
URL: http://svnview.digium.com/svn/asterisk/team/kmoore/channel_event_refactor/channels/chan_iax2.c?view=diff&rev=392857&r1=392856&r2=392857
==============================================================================
--- team/kmoore/channel_event_refactor/channels/chan_iax2.c (original)
+++ team/kmoore/channel_event_refactor/channels/chan_iax2.c Tue Jun 25 13:37:04 2013
@@ -1392,17 +1392,6 @@
reload_config(1);
}
-
-/*! \brief Send manager event at call setup to link between Asterisk channel name
- and IAX2 call identifiers */
-static void iax2_ami_channelupdate(struct chan_iax2_pvt *pvt)
-{
- manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
- "Channel: %s\r\nChanneltype: IAX2\r\nIAX2-callno-local: %d\r\nIAX2-callno-remote: %d\r\nIAX2-peer: %s\r\n",
- pvt->owner ? ast_channel_name(pvt->owner) : "",
- pvt->callno, pvt->peercallno, pvt->peer ? pvt->peer : "");
-}
-
static const struct ast_datastore_info iax2_variable_datastore_info = {
.type = "IAX2_VARIABLE",
.duplicate = iax2_dup_variable_datastore,
@@ -5552,10 +5541,6 @@
{
unsigned short callno = PTR_TO_CALLNO(ast_channel_tech_pvt(c));
ast_debug(1, "Answering IAX2 call\n");
- ast_mutex_lock(&iaxsl[callno]);
- if (iaxs[callno])
- iax2_ami_channelupdate(iaxs[callno]);
- ast_mutex_unlock(&iaxsl[callno]);
return send_command_locked(callno, AST_FRAME_CONTROL, AST_CONTROL_ANSWER, 0, NULL, 0, -1);
}
@@ -5678,7 +5663,6 @@
}
return NULL;
}
- iax2_ami_channelupdate(i);
if (!tmp) {
return NULL;
}
Modified: team/kmoore/channel_event_refactor/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/kmoore/channel_event_refactor/channels/chan_sip.c?view=diff&rev=392857&r1=392856&r2=392857
==============================================================================
--- team/kmoore/channel_event_refactor/channels/chan_sip.c (original)
+++ team/kmoore/channel_event_refactor/channels/chan_sip.c Tue Jun 25 13:37:04 2013
@@ -8249,13 +8249,6 @@
append_history(i, "NewChan", "Channel %s - from %s", ast_channel_name(tmp), i->callid);
}
- /* Inform manager user about new channel and their SIP call ID */
- if (sip_cfg.callevents) {
- manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
- "Channel: %s\r\nUniqueid: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\n",
- ast_channel_name(tmp), ast_channel_uniqueid(tmp), "SIP", i->callid, i->fullcontact);
- }
-
return tmp;
}
@@ -20913,7 +20906,6 @@
ast_cli(a->fd, " From: Domain: %s\n", default_fromdomain);
}
ast_cli(a->fd, " Record SIP history: %s\n", AST_CLI_ONOFF(recordhistory));
- ast_cli(a->fd, " Call Events: %s\n", AST_CLI_ONOFF(sip_cfg.callevents));
ast_cli(a->fd, " Auth. Failure Events: %s\n", AST_CLI_ONOFF(global_authfailureevents));
ast_cli(a->fd, " T.38 support: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT)));
@@ -23166,11 +23158,6 @@
if (!req->ignore && p->owner) {
if (!reinvite) {
ast_queue_control(p->owner, AST_CONTROL_ANSWER);
- if (sip_cfg.callevents) {
- manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
- "Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
- ast_channel_name(p->owner), "SIP", ast_channel_uniqueid(p->owner), p->callid, p->fullcontact, p->peername);
- }
} else { /* RE-invite */
if (p->t38.state == T38_DISABLED || p->t38.state == T38_REJECTED) {
ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
@@ -29607,10 +29594,6 @@
callid = ast_callid_unref(callid);
}
- if (sip_cfg.callevents)
- manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
- "Channel: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
- p->owner? ast_channel_name(p->owner) : "", "SIP", p->callid, p->fullcontact, p->peername);
sip_pvt_unlock(p);
if (!tmpc) {
dialog_unlink_all(p);
@@ -31216,7 +31199,6 @@
/* Misc settings for the channel */
global_relaxdtmf = FALSE;
- sip_cfg.callevents = DEFAULT_CALLEVENTS;
global_authfailureevents = FALSE;
global_t1 = DEFAULT_TIMER_T1;
global_timer_b = 64 * DEFAULT_TIMER_T1;
@@ -31697,8 +31679,6 @@
ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
global_qualifyfreq = DEFAULT_QUALIFYFREQ;
}
- } else if (!strcasecmp(v->name, "callevents")) {
- sip_cfg.callevents = ast_true(v->value);
} else if (!strcasecmp(v->name, "authfailureevents")) {
global_authfailureevents = ast_true(v->value);
} else if (!strcasecmp(v->name, "maxcallbitrate")) {
Modified: team/kmoore/channel_event_refactor/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/team/kmoore/channel_event_refactor/channels/sip/include/sip.h?view=diff&rev=392857&r1=392856&r2=392857
==============================================================================
--- team/kmoore/channel_event_refactor/channels/sip/include/sip.h (original)
+++ team/kmoore/channel_event_refactor/channels/sip/include/sip.h Tue Jun 25 13:37:04 2013
@@ -219,7 +219,6 @@
#define DEFAULT_QUALIFY FALSE /*!< Don't monitor devices */
#define DEFAULT_KEEPALIVE 0 /*!< Don't send keep alive packets */
#define DEFAULT_KEEPALIVE_INTERVAL 60 /*!< Send keep alive packets at 60 second intervals */
-#define DEFAULT_CALLEVENTS FALSE /*!< Extra manager SIP call events */
#define DEFAULT_ALWAYSAUTHREJECT TRUE /*!< Don't reject authentication requests always */
#define DEFAULT_AUTH_OPTIONS FALSE
#define DEFAULT_AUTH_MESSAGE TRUE
@@ -744,7 +743,6 @@
int accept_outofcall_message; /*!< Accept MESSAGE outside of a call */
int compactheaders; /*!< send compact sip headers */
int allow_external_domains; /*!< Accept calls to external SIP domains? */
- int callevents; /*!< Whether we send manager events or not */
int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
int legacy_useroption_parsing; /*!< Whether to strip useroptions in URI via semicolons */
int send_diversion; /*!< Whether to Send SIP Diversion headers */
Modified: team/kmoore/channel_event_refactor/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/kmoore/channel_event_refactor/configs/sip.conf.sample?view=diff&rev=392857&r1=392856&r2=392857
==============================================================================
--- team/kmoore/channel_event_refactor/configs/sip.conf.sample (original)
+++ team/kmoore/channel_event_refactor/configs/sip.conf.sample Tue Jun 25 13:37:04 2013
@@ -397,8 +397,6 @@
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
; for peers and users as well
-;callevents=no ; generate manager events when sip ua
- ; performs events (e.g. hold)
;authfailureevents=no ; generate manager "peerstatus" events when peer can't
; authenticate with Asterisk. Peerstatus will be "rejected".
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
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