[svn-commits] russell: trunk r57365 - in /trunk: ./ apps/ configs/
doc/ include/asterisk/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Thu Mar 1 16:44:10 MST 2007
Author: russell
Date: Thu Mar 1 17:44:09 2007
New Revision: 57365
URL: http://svn.digium.com/view/asterisk?view=rev&rev=57365
Log:
Merged revisions 57364 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines
Merge changes from svn/asterisk/team/russell/sla_updates
* Originally, I put in the documentation that only Zap interfaces would be
supported on the trunk side. However, after a discussion with Qwell, we came
up with a way to make IP trunks work as well, using some things already in
Asterisk. So, here it is, this now officially supports IP trunks.
* Update the SLA documentation to reflect how to setup IP trunks.
* Add a section in sla.txt that describes how to set up an SLA system with
voicemail.
* Simplify the way DTMF passthrough is handled in MeetMe.
* Fix a bug that exposed itself when using a Local channel on the trunk side
in SLA. The station's channel needs to be passed to the dial API when
dialing the trunk.
* Change a WARNING message to DEBUG in channel.h. This message is of no use
to users.
........
Modified:
trunk/ (props changed)
trunk/apps/app_meetme.c
trunk/configs/sla.conf.sample
trunk/doc/sla.txt
trunk/include/asterisk/channel.h
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_meetme.c?view=diff&rev=57365&r1=57364&r2=57365
==============================================================================
--- trunk/apps/app_meetme.c (original)
+++ trunk/apps/app_meetme.c Thu Mar 1 17:44:09 2007
@@ -347,7 +347,6 @@
time_t jointime; /*!< Time the user joined the conference */
struct volume talk;
struct volume listen;
- AST_LIST_HEAD_NOLOCK(, ast_frame) frame_q;
AST_LIST_ENTRY(ast_conf_user) list;
};
@@ -1238,17 +1237,15 @@
}
static void conf_queue_dtmf(const struct ast_conference *conf,
- const struct ast_conf_user *sender, const struct ast_frame *_f)
-{
- struct ast_frame *f;
+ const struct ast_conf_user *sender, struct ast_frame *f)
+{
struct ast_conf_user *user;
AST_LIST_TRAVERSE(&conf->userlist, user, list) {
if (user == sender)
continue;
- if (!(f = ast_frdup(_f)))
- return;
- AST_LIST_INSERT_TAIL(&user->frame_q, f, frame_list);
+ if (ast_write(user->chan, f) < 0)
+ ast_log(LOG_WARNING, "Error writing frame to channel %s\n", user->chan->name);
}
}
@@ -1875,14 +1872,6 @@
f = ast_read(c);
if (!f)
break;
- if (!AST_LIST_EMPTY(&user->frame_q)) {
- struct ast_frame *f;
- f = AST_LIST_REMOVE_HEAD(&user->frame_q, frame_list);
- if (ast_write(chan, f) < 0) {
- ast_log(LOG_WARNING, "Error writing frame to channel!\n");
- }
- ast_frfree(f);
- }
if ((f->frametype == AST_FRAME_VOICE) && (f->subclass == AST_FORMAT_SLINEAR)) {
if (user->talk.actual)
ast_frame_adjust_volume(f, user->talk.actual);
@@ -3941,7 +3930,7 @@
}
struct dial_trunk_args {
- struct sla_trunk *trunk;
+ struct sla_trunk_ref *trunk_ref;
struct sla_station *station;
ast_mutex_t *cond_lock;
ast_cond_t *cond;
@@ -3956,7 +3945,7 @@
char conf_name[MAX_CONFNUM];
struct ast_conference *conf;
struct ast_flags conf_flags = { 0 };
- struct sla_trunk *trunk = args->trunk;
+ struct sla_trunk_ref *trunk_ref = args->trunk_ref;
if (!(dial = ast_dial_create())) {
ast_mutex_lock(args->cond_lock);
@@ -3965,7 +3954,7 @@
return NULL;
}
- tech_data = ast_strdupa(trunk->device);
+ tech_data = ast_strdupa(trunk_ref->trunk->device);
tech = strsep(&tech_data, "/");
if (ast_dial_append(dial, tech, tech_data) == -1) {
ast_mutex_lock(args->cond_lock);
@@ -3975,7 +3964,7 @@
return NULL;
}
- dial_res = ast_dial_run(dial, NULL, 1);
+ dial_res = ast_dial_run(dial, trunk_ref->chan, 1);
if (dial_res != AST_DIAL_RESULT_TRYING) {
ast_mutex_lock(args->cond_lock);
ast_cond_signal(args->cond);
@@ -3988,7 +3977,7 @@
unsigned int done = 0;
switch ((dial_res = ast_dial_state(dial))) {
case AST_DIAL_RESULT_ANSWERED:
- trunk->chan = ast_dial_answered(dial);
+ trunk_ref->trunk->chan = ast_dial_answered(dial);
case AST_DIAL_RESULT_HANGUP:
case AST_DIAL_RESULT_INVALID:
case AST_DIAL_RESULT_FAILED:
@@ -4005,7 +3994,7 @@
break;
}
- if (!trunk->chan) {
+ if (!trunk_ref->trunk->chan) {
ast_mutex_lock(args->cond_lock);
ast_cond_signal(args->cond);
ast_mutex_unlock(args->cond_lock);
@@ -4014,7 +4003,7 @@
return NULL;
}
- snprintf(conf_name, sizeof(conf_name), "SLA_%s", trunk->name);
+ snprintf(conf_name, sizeof(conf_name), "SLA_%s", trunk_ref->trunk->name);
ast_set_flag(&conf_flags,
CONFFLAG_QUIET | CONFFLAG_MARKEDEXIT | CONFFLAG_MARKEDUSER |
CONFFLAG_PASS_DTMF | CONFFLAG_SLA_TRUNK);
@@ -4025,12 +4014,12 @@
ast_mutex_unlock(args->cond_lock);
if (conf) {
- conf_run(trunk->chan, conf, conf_flags.flags, NULL);
+ conf_run(trunk_ref->trunk->chan, conf, conf_flags.flags, NULL);
dispose_conf(conf);
conf = NULL;
}
- trunk->chan = NULL;
+ trunk_ref->trunk->chan = NULL;
ast_dial_join(dial);
ast_dial_destroy(dial);
@@ -4105,13 +4094,15 @@
return 0;
}
+ trunk_ref->chan = chan;
+
if (!trunk_ref->trunk->chan) {
ast_mutex_t cond_lock;
ast_cond_t cond;
pthread_t dont_care;
pthread_attr_t attr;
struct dial_trunk_args args = {
- .trunk = trunk_ref->trunk,
+ .trunk_ref = trunk_ref,
.station = station,
.cond_lock = &cond_lock,
.cond = &cond,
@@ -4137,6 +4128,7 @@
ast_log(LOG_DEBUG, "Trunk didn't get created. chan: %lx\n", (long) trunk_ref->trunk->chan);
pbx_builtin_setvar_helper(chan, "SLASTATION_STATUS", "CONGESTION");
sla_change_trunk_state(trunk_ref->trunk, SLA_TRUNK_STATE_IDLE, ALL_TRUNK_REFS);
+ trunk_ref->chan = NULL;
return 0;
}
}
@@ -4145,7 +4137,6 @@
snprintf(conf_name, sizeof(conf_name), "SLA_%s", trunk_ref->trunk->name);
ast_set_flag(&conf_flags,
CONFFLAG_QUIET | CONFFLAG_MARKEDEXIT | CONFFLAG_PASS_DTMF | CONFFLAG_SLA_STATION);
- trunk_ref->chan = chan;
ast_answer(chan);
conf = build_conf(conf_name, "", "", 0, 0, 1);
if (conf) {
Modified: trunk/configs/sla.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sla.conf.sample?view=diff&rev=57365&r1=57364&r2=57365
==============================================================================
--- trunk/configs/sla.conf.sample (original)
+++ trunk/configs/sla.conf.sample Thu Mar 1 17:44:09 2007
@@ -18,7 +18,10 @@
;type=trunk ; This line is what marks this entry as a trunk.
;device=Zap/3 ; Map this trunk declaration to a specific device.
- ; NOTE: At this point, this *must* be a zap channel!
+ ; NOTE: You can not just put any type of channel here.
+ ; Zap channels can be directly used. IP trunks
+ ; require some indirect configuration which is
+ ; described in doc/sla.txt.
;autocontext=line1 ; This supports automatic generation of the dialplan entries
; if the autocontext option is used. Each trunk should have
@@ -52,7 +55,10 @@
;[line4]
;type=trunk
-;device=Zap/4
+;device=Local/disa at line4_outbound ; A Local channel in combination with the Disa
+ ; application can be used to support IP trunks.
+ ; See doc/sla.txt on more information on how
+ ; IP trunks work.
;autocontext=line4
; --------------------------------------
Modified: trunk/doc/sla.txt
URL: http://svn.digium.com/view/asterisk/trunk/doc/sla.txt?view=diff&rev=57365&r1=57364&r2=57365
==============================================================================
--- trunk/doc/sla.txt (original)
+++ trunk/doc/sla.txt Thu Mar 1 17:44:09 2007
@@ -18,7 +18,10 @@
basic operation if the "autocontext" option is set for trunks and stations in
sla.conf. However, for reference, here is an automatically generated dialplan
to help with custom building of the dialplan to include other features, such as
-voicemail:
+voicemail.
+
+However, note that there is a little bit of additional configuration needed if
+the trunk is an IP channel. This is discussed in the TRUNKS section.
[line1]
exten => s,1,SLATrunk(line1)
@@ -49,14 +52,52 @@
-------------------------------------------------------------------------------
TRUNKS
-For the trunk side of SLA, the only channels that are currently supported are
-Zap channels. Support for IP trunks is planned, but not yet implemented.
-
Be sure to configure the trunk's context to be the same one that is set for the
"autocontext" option in sla.conf if automatic dialplan configuration is used.
+This would be done in the regular device entry in zapata.conf, sip.conf, etc.
+Note that the automatic dialplan generation creates the SLATrunk() extension
+at extension 's'. This is perfect for Zap channels that are FXO trunks, for
+example. However, it may not be good enough for an IP trunk, since the call
+coming in over the trunk may specify an actual number.
If the dialplan is being built manually, ensure that calls coming in on a trunk
-execute the SLATrunk() application with an argument of the trunk name.
+execute the SLATrunk() application with an argument of the trunk name, as shown
+in the dialplan example before.
+
+IP trunks can be used, but they require some additional configuration to work.
+
+For this example, let's say we have a SIP trunk called "mytrunk" that is going
+to be used as line4. Furthermore, when calls come in on this trunk, they are
+going to say that they are calling the number "12564286000". Also, let's say
+that the numbers that are valid for calling out this trunk are NANP numbers,
+of the form _1NXXNXXXXXX.
+
+In sip.conf, there would be an entry for [mytrunk]. For [mytrunk],
+set context=line4.
+
+
+sla.conf:
+
+[line4]
+type=trunk
+device=Local/disa at line4_outbound
+
+
+extensions.conf:
+
+[line4]
+exten => 12564286000,1,SLATrunk(line4)
+
+[line4_outbound]
+exten => disa,1,Disa(no-password|line4_outbound)
+exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@mytrunk)
+
+
+So, when a station picks up their phone and connects to line 4, they are
+connected to the local dialplan. The Disa application plays dialtone to the
+phone and collects digits until it matches an extension. In this case, once
+the phone dials a number like 12565551212, the call will proceed out the
+SIP trunk.
-------------------------------------------------------------------------------
@@ -91,3 +132,94 @@
is taken off hook, then the phone should be automatically configured to
dial "station1" when it is taken off hook.
-------------------------------------------------------------------------------
+
+
+-------------------------------------------------------------------------------
+VOICEMAIL
+
+This is an example of how you could set up a single voicemail box for the
+phone system. The voicemail box number used in this example is 1234, which
+would be configured in voicemail.conf.
+
+For this example, assume that there are 2 trunks and 3 stations. The trunks
+are Zap/1 and Zap/2. The stations are SIP/station1, SIP/station2, and
+SIP/station3.
+
+In zapata.conf, channel 1 has context=line1 and channel 2 has context=line2.
+
+In sip.conf, all three stations are configured with context=sla_stations.
+
+When the stations pick up their phones to dial, they are allowed to dial
+NANP numbers for outbound calls, or 8500 for checking voicemail.
+
+
+sla.conf:
+
+[line1]
+type=trunk
+device=Local/disa at line1_outbound
+
+[line2]
+type=trunk
+device=Local/disa at line2_outbound
+
+[station](!)
+type=station
+trunk=line1
+trunk=line2
+
+[station1](station)
+device=SIP/station1
+
+[station2](station)
+device=SIP/station2
+
+[station3](station)
+device=SIP/station3
+
+
+extensions.conf:
+
+[macro-slaline]
+exten => s,1,SLATrunk(${ARG1})
+exten => s,n,Goto(s-${SLATRUNK_STATUS}|1)
+exten => s-FAILURE,1,Voicemail(1234|u)
+exten => s-UNANSWERED,1,Voicemail(1234|u)
+
+[line1]
+exten => s,1,Macro(slaline|line1)
+
+[line2]
+exten => s,2,Macro(slaline|line2)
+
+[line1_outbound]
+exten => disa,1,Disa(no-password|line1_outbound)
+exten => _1NXXNXXXXXX,1,Dial(Zap/1/${EXTEN})
+exten => 8500,1,VoicemailMain(1234)
+
+[line2_outbound]
+exten => disa,1,Disa(no-password|line2_outbound)
+exten => _1NXXNXXXXXX,1,Dial(Zap/2/${EXTEN})
+exten => 8500,1,VoicemailMain(1234)
+
+[sla_stations]
+
+exten => station1,1,SLAStation(station1)
+exten => station1_line1,hint,SLA:station1_line1
+exten => station1_line1,1,SLAStation(station1_line1)
+exten => station1_line2,hint,SLA:station1_line2
+exten => station1_line2,1,SLAStation(station1_line2)
+
+exten => station2,1,SLAStation(station2)
+exten => station2_line1,hint,SLA:station2_line1
+exten => station2_line1,1,SLAStation(station2_line1)
+exten => station2_line2,hint,SLA:station2_line2
+exten => station2_line2,1,SLAStation(station2_line2)
+
+exten => station3,1,SLAStation(station3)
+exten => station3_line1,hint,SLA:station3_line1
+exten => station3_line1,1,SLAStation(station3_line1)
+exten => station3_line2,hint,SLA:station3_line2
+exten => station3_line2,1,SLAStation(station3_line2)
+
+-------------------------------------------------------------------------------
Modified: trunk/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/channel.h?view=diff&rev=57365&r1=57364&r2=57365
==============================================================================
--- trunk/include/asterisk/channel.h (original)
+++ trunk/include/asterisk/channel.h Thu Mar 1 17:44:09 2007
@@ -1375,15 +1375,16 @@
#define CRASH do { } while(0)
#endif
-#define CHECK_BLOCKING(c) { \
- if (ast_test_flag(c, AST_FLAG_BLOCKING)) {\
- ast_log(LOG_WARNING, "Thread %ld Blocking '%s', already blocked by thread %ld in procedure %s\n", (long) pthread_self(), (c)->name, (long) (c)->blocker, (c)->blockproc); \
- CRASH; \
- } else { \
- (c)->blocker = pthread_self(); \
- (c)->blockproc = __PRETTY_FUNCTION__; \
- ast_set_flag(c, AST_FLAG_BLOCKING); \
- } }
+#define CHECK_BLOCKING(c) do { \
+ if (ast_test_flag(c, AST_FLAG_BLOCKING)) {\
+ if (option_debug) \
+ ast_log(LOG_DEBUG, "Thread %ld Blocking '%s', already blocked by thread %ld in procedure %s\n", (long) pthread_self(), (c)->name, (long) (c)->blocker, (c)->blockproc); \
+ CRASH; \
+ } else { \
+ (c)->blocker = pthread_self(); \
+ (c)->blockproc = __PRETTY_FUNCTION__; \
+ ast_set_flag(c, AST_FLAG_BLOCKING); \
+ } } while (0)
ast_group_t ast_get_group(const char *s);
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