Merged revisions 53810 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines

Merge team/russell/sla_rewrite

This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4.  It is now functional and ready for testing.  However, I will be
adding some additional features over the next week, as well.

For information on how to set this up, see configs/sla.conf.sample 
and doc/sla.txt.

In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:

chan_sip:
 - Add the ability to indicate HOLD state in NOTIFY messages.
 - Queue HOLD and UNHOLD control frames even if the channel is not bridged to
   another channel.

linkedlists.h:
 - Add support for rwlock based linked lists.

dial.c:
 - Add the ability to run ast_dial_start() without a reference channel to
   inherit information from.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53817 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -4818,7 +4818,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 	int iterator;
 	int sendonly = 0;
 	int numberofports;
-	struct ast_channel *bridgepeer = NULL;
 	struct ast_rtp *newaudiortp, *newvideortp;	/* Buffers for codec handling */
 	int newjointcapability;				/* Negotiated capability */
 	int newpeercapability;
@@ -5309,13 +5308,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 		ast_set_write_format(p->owner, p->owner->writeformat);
 	}
 	
-	/* Turn on/off music on hold if we are holding/unholding */
-	if ((bridgepeer = ast_bridged_channel(p->owner))) {
 	if (sin.sin_addr.s_addr && !sendonly) {
+		ast_log(LOG_DEBUG, "Queueing UNHOLD!\n");
 		ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
 		/* Activate a re-invite */
 		ast_queue_frame(p->owner, &ast_null_frame);
 	} else if (!sin.sin_addr.s_addr || sendonly) {
+		ast_log(LOG_DEBUG, "Going on HOLD!\n");
 		ast_queue_control_data(p->owner, AST_CONTROL_HOLD, 
 				       S_OR(p->mohsuggest, NULL),
 				       !ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
@@ -5325,7 +5324,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 		/* Activate a re-invite */
 		ast_queue_frame(p->owner, &ast_null_frame);
 	}
-	}
 
 	/* Manager Hold and Unhold events must be generated, if necessary */
 	if (sin.sin_addr.s_addr && !sendonly) {

configs/sla.conf.sample

@@ -1,26 +1,82 @@
-; Configuration file for SLAs (Shared Line Appearances).
-
-; Defining a SLA uses the following syntax:
 ;
-; type => Technology/TechData
+; Configuration for Shared Line Appearances (SLA).
-;
-;    type       => trunk or station
-;    Technology => the channel driver (see show channeltypes)
-;    TechData   => the data specific to the channel driver
-;
-; The Technology/TechData information is the same as that provided to the
-; Dial application.
 ;
 
-; define a SLA called junky
+; ---- General Options ----------------
-[junky]
+[general]
-trunk => SIP/10
+; There are none!
-station => SIP/15
-station => SIP/16
 
-;define a SLA called markster
+; -------------------------------------
-[markster]
+
-trunk => Zap/1
+
-station => SIP/20
+; ---- Trunk Declarations -------------
+
+; Provide a name for this trunk.
+[line1]
+; This line is what marks this entry as a trunk.
+type=trunk
+; Map this trunk declaration to a specific device.
+; NOTE: At this point, this *must* be a zap channel!
+device=Zap/3
+; This supports automatic generation of the dialplan entries if the autocontext
+; option is used.  Each trunk should have a unique context name.  Then, in
+; zapata.conf, this device should be configured to have incoming calls go to
+; this context.
+autocontext=line1
+
+[line2]
+type=trunk
+device=Zap/4
+autocontext=line2
+
+[line3]
+type=trunk
+device=Zap/3
+autocontext=line3
+
+[line4]
+type=trunk
+device=Zap/4
+autocontext=line4
+; --------------------------------------
+
+
+; ---- Station Declarations ------------
+
+; In this example, all stations have the same trunks, so this configuration
+; template is used to simplify the declaration of each station.
+[station](!)
+; This line indicates that this entry is a station.
+type=station
+; This supports automatic generation of the dialplan entries if the autocontext
+; option is used.  All stations can use the same context without conflict.  The
+; device for this station should have its context configured to the same one
+; listed here.
+autocontext=sla_stations
+; Individually list all of the trunks that will appear on this station.  This
+; order is significant.  It should be the same order as they appear on the
+; phone.  The order here defines the order of preference that the trunks will
+; be used.
+trunk=line1
+trunk=line2
+trunk=line3
+trunk=line4
+
+; Define a station that uses the configuration from the template "station".
+[station1](station)
+; Each station must be mapped to a device.
+device=SIP/station1
+
+[station2](station)
+device=SIP/station2
+
+[station3](station)
+device=SIP/station3
+
+[station4](station)
+device=SIP/station4
+
+[station5](station)
+device=SIP/station5
+; --------------------------------------
 
-; Also you can see SLA infos via the CLI, by typing "sla show"

doc/sla.txt

@@ -0,0 +1,93 @@
+-------------------------------------------------------------
+--- Shared Line Appearances ---------------------------------
+-------------------------------------------------------------
+
+-------------------------------------------------------------
+INTRODUCTION
+
+The "SLA" functionality in Asterisk is intended to allow a setup that emulates
+a simple key system.  It uses the various abstraction layers already built into
+Asterisk to emulate key system functionality across various devices, including
+IP channels.
+-------------------------------------------------------------
+
+-------------------------------------------------------------
+DIALPLAN CONFIGURATION
+
+The SLA implementation can automatically generate the dialplan necessary for
+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:
+
+[line1]
+exten => s,1,SLATrunk(line1)
+
+[line2]
+exten => s,2,SLATRUNK(line2)
+
+[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)
+-------------------------------------------------------------
+
+-------------------------------------------------------------
+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.
+
+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.
+-------------------------------------------------------------
+
+
+-------------------------------------------------------------
+STATIONS
+
+Currently, the only channel driver that has all of the features necessary to
+support an SLA environment is chan_sip.  Here are some hints on configuring
+a SIP phone for use with SLA:
+
+1) Add the SIP channel as a [station] in sla.conf.
+
+2) Configure the phone in sip.conf.  If automatic dialplan configuration was
+   used by enabling the "autocontext" option in sla.conf, then this entry in
+   sip.conf should have the same context setting.
+
+3) On the phone itself, there are various things that must be configured to
+   make everything work correctly:
+
+   Let's say this phone is called "station1" in sla.conf, and it uses trunks
+   named "line1" and line2".
+
+   a) Two line buttons must be configured to subscribe to the state of the
+      following extensions:
+        - station1_line1
+        - station1_line2
+
+   b) The line appearance buttons should be configured to dial the extensions
+      that they are subscribed to when they are pressed.
+
+   c) If you would like the phone to automatically connect to a trunk when it
+      is taken off hook, then the phone should be automatically configured to
+      dial "station1" when it is taken off hook.
+-------------------------------------------------------------

include/asterisk/app.h

@@ -308,6 +308,9 @@ struct ast_app_option {
 	unsigned int arg_index;
 };
 
+#define BEGIN_OPTIONS {
+#define END_OPTIONS }
+
 /*!
   \brief Declares an array of options for an application.
   \param holder The name of the array to be created

include/asterisk/dial.h

@@ -42,7 +42,7 @@ enum ast_dial_option {
 
 /*! \brief List of return codes for dial run API calls */
 enum ast_dial_result {
-	AST_DIAL_RESULT_INVALID = 0, /*!< Invalid options were passed to run function */
+	AST_DIAL_RESULT_INVALID,     /*!< Invalid options were passed to run function */
 	AST_DIAL_RESULT_FAILED,      /*!< Attempts to dial failed before reaching critical state */
 	AST_DIAL_RESULT_TRYING,      /*!< Currently trying to dial */
 	AST_DIAL_RESULT_RINGING,     /*!< Dial is presently ringing */

include/asterisk/utils.h

@@ -553,5 +553,8 @@ int _ast_vasprintf(char **ret, const char *file, int lineno, const char *func, c
  */
 void ast_enable_packet_fragmentation(int sock);
 
+#define ARRAY_LEN(a) (sizeof(a) / sizeof(a[0]))
+
 #include "asterisk/strings.h"
+
 #endif /* _ASTERISK_UTILS_H */

main/dial.c

@@ -228,14 +228,17 @@ static int begin_dial(struct ast_dial *dial, struct ast_channel *chan)
 		ast_copy_string(numsubst, channel->device, sizeof(numsubst));
 
 		/* Request that the channel be created */
-		if (!(channel->owner = ast_request(channel->tech, chan->nativeformats, numsubst, &channel->cause)))
+		if (!(channel->owner = ast_request(channel->tech, 
+			chan ? chan->nativeformats : AST_FORMAT_AUDIO_MASK, numsubst, &channel->cause))) {
 			continue;
+		}
 
 		channel->owner->appl = "AppDial2";
                 channel->owner->data = "(Outgoing Line)";
                 channel->owner->whentohangup = 0;
 
 		/* Inherit everything from he who spawned this Dial */
+		if (chan) {
 			ast_channel_inherit_variables(chan, channel->owner);
 
 			/* Copy over callerid information */
@@ -255,6 +258,7 @@ static int begin_dial(struct ast_dial *dial, struct ast_channel *chan)
 			channel->owner->cid.cid_tns = chan->cid.cid_tns;
 			channel->owner->adsicpe = chan->adsicpe;
 			channel->owner->transfercapability = chan->transfercapability;
+		}
 
 		/* Actually call the device */
 		if ((res = ast_call(channel->owner, numsubst, 0))) {
@@ -530,12 +534,16 @@ enum ast_dial_result ast_dial_run(struct ast_dial *dial, struct ast_channel *cha
 	enum ast_dial_result res = AST_DIAL_RESULT_TRYING;
 
 	/* Ensure required arguments are passed */
-	if (!dial || !chan)
+	if (!dial || (!chan && !async)) {
+		ast_log(LOG_DEBUG, "invalid #1\n");
 		return AST_DIAL_RESULT_INVALID;
+	}
 
 	/* If there are no channels to dial we can't very well try to dial them */
-	if (AST_LIST_EMPTY(&dial->channels))
+	if (AST_LIST_EMPTY(&dial->channels)) {
+		ast_log(LOG_DEBUG, "invalid #2\n");
 		return AST_DIAL_RESULT_INVALID;
+	}
 
 	/* Dial each requested channel */
 	if (!begin_dial(dial, chan))
@@ -543,6 +551,7 @@ enum ast_dial_result ast_dial_run(struct ast_dial *dial, struct ast_channel *cha
 
 	/* If we are running async spawn a thread and send it away... otherwise block here */
 	if (async) {
+		dial->status = AST_DIAL_RESULT_TRYING;
 		/* Try to create a thread */
 		if (ast_pthread_create(&dial->thread, NULL, async_dial, dial)) {
 			/* Failed to create the thread - hangup all dialed channels and return failed */