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: http://svn.digium.com/svn/asterisk/trunk@53817 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
russell
parent
0fbac396a5
commit
be94f38009
1842
apps/app_meetme.c
1842
apps/app_meetme.c
File diff suppressed because it is too large
Load Diff
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@ -4818,7 +4818,6 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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int iterator;
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int sendonly = 0;
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int numberofports;
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struct ast_channel *bridgepeer = NULL;
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struct ast_rtp *newaudiortp, *newvideortp; /* Buffers for codec handling */
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int newjointcapability; /* Negotiated capability */
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int newpeercapability;
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@ -5309,22 +5308,21 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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ast_set_write_format(p->owner, p->owner->writeformat);
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}
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/* Turn on/off music on hold if we are holding/unholding */
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if ((bridgepeer = ast_bridged_channel(p->owner))) {
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if (sin.sin_addr.s_addr && !sendonly) {
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ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
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/* Activate a re-invite */
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ast_queue_frame(p->owner, &ast_null_frame);
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} else if (!sin.sin_addr.s_addr || sendonly) {
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ast_queue_control_data(p->owner, AST_CONTROL_HOLD,
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S_OR(p->mohsuggest, NULL),
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!ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
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if (sendonly)
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ast_rtp_stop(p->rtp);
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/* RTCP needs to go ahead, even if we're on hold!!! */
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/* Activate a re-invite */
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ast_queue_frame(p->owner, &ast_null_frame);
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}
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if (sin.sin_addr.s_addr && !sendonly) {
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ast_log(LOG_DEBUG, "Queueing UNHOLD!\n");
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ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
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/* Activate a re-invite */
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ast_queue_frame(p->owner, &ast_null_frame);
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} else if (!sin.sin_addr.s_addr || sendonly) {
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ast_log(LOG_DEBUG, "Going on HOLD!\n");
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ast_queue_control_data(p->owner, AST_CONTROL_HOLD,
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S_OR(p->mohsuggest, NULL),
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!ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
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if (sendonly)
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ast_rtp_stop(p->rtp);
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/* RTCP needs to go ahead, even if we're on hold!!! */
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/* Activate a re-invite */
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ast_queue_frame(p->owner, &ast_null_frame);
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}
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/* Manager Hold and Unhold events must be generated, if necessary */
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@ -1,26 +1,82 @@
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; Configuration file for SLAs (Shared Line Appearances).
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; Defining a SLA uses the following syntax:
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;
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; type => Technology/TechData
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;
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; type => trunk or station
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; Technology => the channel driver (see show channeltypes)
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; TechData => the data specific to the channel driver
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;
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; The Technology/TechData information is the same as that provided to the
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; Dial application.
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; Configuration for Shared Line Appearances (SLA).
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;
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; define a SLA called junky
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[junky]
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trunk => SIP/10
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station => SIP/15
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station => SIP/16
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; ---- General Options ----------------
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[general]
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; There are none!
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;define a SLA called markster
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[markster]
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trunk => Zap/1
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station => SIP/20
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; -------------------------------------
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; ---- Trunk Declarations -------------
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; Provide a name for this trunk.
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[line1]
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; This line is what marks this entry as a trunk.
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type=trunk
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; Map this trunk declaration to a specific device.
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; NOTE: At this point, this *must* be a zap channel!
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device=Zap/3
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; This supports automatic generation of the dialplan entries if the autocontext
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; option is used. Each trunk should have a unique context name. Then, in
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; zapata.conf, this device should be configured to have incoming calls go to
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; this context.
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autocontext=line1
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[line2]
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type=trunk
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device=Zap/4
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autocontext=line2
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[line3]
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type=trunk
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device=Zap/3
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autocontext=line3
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[line4]
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type=trunk
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device=Zap/4
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autocontext=line4
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; --------------------------------------
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; ---- Station Declarations ------------
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; In this example, all stations have the same trunks, so this configuration
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; template is used to simplify the declaration of each station.
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[station](!)
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; This line indicates that this entry is a station.
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type=station
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; This supports automatic generation of the dialplan entries if the autocontext
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; option is used. All stations can use the same context without conflict. The
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; device for this station should have its context configured to the same one
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; listed here.
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autocontext=sla_stations
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; Individually list all of the trunks that will appear on this station. This
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; order is significant. It should be the same order as they appear on the
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; phone. The order here defines the order of preference that the trunks will
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; be used.
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trunk=line1
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trunk=line2
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trunk=line3
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trunk=line4
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; Define a station that uses the configuration from the template "station".
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[station1](station)
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; Each station must be mapped to a device.
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device=SIP/station1
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[station2](station)
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device=SIP/station2
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[station3](station)
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device=SIP/station3
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[station4](station)
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device=SIP/station4
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[station5](station)
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device=SIP/station5
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; --------------------------------------
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; Also you can see SLA infos via the CLI, by typing "sla show"
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@ -0,0 +1,93 @@
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-------------------------------------------------------------
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--- Shared Line Appearances ---------------------------------
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-------------------------------------------------------------
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-------------------------------------------------------------
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INTRODUCTION
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The "SLA" functionality in Asterisk is intended to allow a setup that emulates
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a simple key system. It uses the various abstraction layers already built into
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Asterisk to emulate key system functionality across various devices, including
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IP channels.
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-------------------------------------------------------------
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-------------------------------------------------------------
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DIALPLAN CONFIGURATION
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The SLA implementation can automatically generate the dialplan necessary for
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basic operation if the "autocontext" option is set for trunks and stations in
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sla.conf. However, for reference, here is an automatically generated dialplan
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to help with custom building of the dialplan to include other features, such as
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voicemail:
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[line1]
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exten => s,1,SLATrunk(line1)
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[line2]
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exten => s,2,SLATRUNK(line2)
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[sla_stations]
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exten => station1,1,SLAStation(station1)
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exten => station1_line1,hint,SLA:station1_line1
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exten => station1_line1,1,SLAStation(station1_line1)
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exten => station1_line2,hint,SLA:station1_line2
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exten => station1_line2,1,SLAStation(station1_line2)
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exten => station2,1,SLAStation(station2)
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exten => station2_line1,hint,SLA:station2_line1
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exten => station2_line1,1,SLAStation(station2_line1)
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exten => station2_line2,hint,SLA:station2_line2
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exten => station2_line2,1,SLAStation(station2_line2)
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exten => station3,1,SLAStation(station3)
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exten => station3_line1,hint,SLA:station3_line1
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exten => station3_line1,1,SLAStation(station3_line1)
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exten => station3_line2,hint,SLA:station3_line2
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exten => station3_line2,1,SLAStation(station3_line2)
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-------------------------------------------------------------
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-------------------------------------------------------------
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TRUNKS
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For the trunk side of SLA, the only channels that are currently supported are
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Zap channels. Support for IP trunks is planned, but not yet implemented.
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Be sure to configure the trunk's context to be the same one that is set for the
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"autocontext" option in sla.conf if automatic dialplan configuration is used.
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If the dialplan is being built manually, ensure that calls coming in on a trunk
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execute the SLATrunk() application with an argument of the trunk name.
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-------------------------------------------------------------
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-------------------------------------------------------------
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STATIONS
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Currently, the only channel driver that has all of the features necessary to
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support an SLA environment is chan_sip. Here are some hints on configuring
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a SIP phone for use with SLA:
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1) Add the SIP channel as a [station] in sla.conf.
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2) Configure the phone in sip.conf. If automatic dialplan configuration was
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used by enabling the "autocontext" option in sla.conf, then this entry in
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sip.conf should have the same context setting.
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3) On the phone itself, there are various things that must be configured to
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make everything work correctly:
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Let's say this phone is called "station1" in sla.conf, and it uses trunks
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named "line1" and line2".
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a) Two line buttons must be configured to subscribe to the state of the
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following extensions:
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- station1_line1
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- station1_line2
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b) The line appearance buttons should be configured to dial the extensions
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that they are subscribed to when they are pressed.
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c) If you would like the phone to automatically connect to a trunk when it
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is taken off hook, then the phone should be automatically configured to
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dial "station1" when it is taken off hook.
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-------------------------------------------------------------
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@ -308,6 +308,9 @@ struct ast_app_option {
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unsigned int arg_index;
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};
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#define BEGIN_OPTIONS {
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#define END_OPTIONS }
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/*!
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\brief Declares an array of options for an application.
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\param holder The name of the array to be created
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@ -42,7 +42,7 @@ enum ast_dial_option {
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/*! \brief List of return codes for dial run API calls */
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enum ast_dial_result {
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AST_DIAL_RESULT_INVALID = 0, /*!< Invalid options were passed to run function */
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AST_DIAL_RESULT_INVALID, /*!< Invalid options were passed to run function */
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AST_DIAL_RESULT_FAILED, /*!< Attempts to dial failed before reaching critical state */
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AST_DIAL_RESULT_TRYING, /*!< Currently trying to dial */
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AST_DIAL_RESULT_RINGING, /*!< Dial is presently ringing */
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@ -553,5 +553,8 @@ int _ast_vasprintf(char **ret, const char *file, int lineno, const char *func, c
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*/
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void ast_enable_packet_fragmentation(int sock);
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#define ARRAY_LEN(a) (sizeof(a) / sizeof(a[0]))
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#include "asterisk/strings.h"
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#endif /* _ASTERISK_UTILS_H */
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51
main/dial.c
51
main/dial.c
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@ -228,33 +228,37 @@ static int begin_dial(struct ast_dial *dial, struct ast_channel *chan)
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ast_copy_string(numsubst, channel->device, sizeof(numsubst));
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/* Request that the channel be created */
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if (!(channel->owner = ast_request(channel->tech, chan->nativeformats, numsubst, &channel->cause)))
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if (!(channel->owner = ast_request(channel->tech,
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chan ? chan->nativeformats : AST_FORMAT_AUDIO_MASK, numsubst, &channel->cause))) {
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continue;
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}
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channel->owner->appl = "AppDial2";
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channel->owner->data = "(Outgoing Line)";
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channel->owner->whentohangup = 0;
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/* Inherit everything from he who spawned this Dial */
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ast_channel_inherit_variables(chan, channel->owner);
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if (chan) {
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ast_channel_inherit_variables(chan, channel->owner);
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/* Copy over callerid information */
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S_REPLACE(channel->owner->cid.cid_num, ast_strdup(chan->cid.cid_num));
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S_REPLACE(channel->owner->cid.cid_name, ast_strdup(chan->cid.cid_name));
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S_REPLACE(channel->owner->cid.cid_ani, ast_strdup(chan->cid.cid_ani));
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S_REPLACE(channel->owner->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
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ast_string_field_set(channel->owner, language, chan->language);
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ast_string_field_set(channel->owner, accountcode, chan->accountcode);
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channel->owner->cdrflags = chan->cdrflags;
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if (ast_strlen_zero(channel->owner->musicclass))
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ast_string_field_set(channel->owner, musicclass, chan->musicclass);
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channel->owner->cid.cid_pres = chan->cid.cid_pres;
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channel->owner->cid.cid_ton = chan->cid.cid_ton;
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channel->owner->cid.cid_tns = chan->cid.cid_tns;
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channel->owner->adsicpe = chan->adsicpe;
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channel->owner->transfercapability = chan->transfercapability;
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/* Copy over callerid information */
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S_REPLACE(channel->owner->cid.cid_num, ast_strdup(chan->cid.cid_num));
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S_REPLACE(channel->owner->cid.cid_name, ast_strdup(chan->cid.cid_name));
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S_REPLACE(channel->owner->cid.cid_ani, ast_strdup(chan->cid.cid_ani));
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S_REPLACE(channel->owner->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
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ast_string_field_set(channel->owner, language, chan->language);
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ast_string_field_set(channel->owner, accountcode, chan->accountcode);
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channel->owner->cdrflags = chan->cdrflags;
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if (ast_strlen_zero(channel->owner->musicclass))
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ast_string_field_set(channel->owner, musicclass, chan->musicclass);
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channel->owner->cid.cid_pres = chan->cid.cid_pres;
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channel->owner->cid.cid_ton = chan->cid.cid_ton;
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channel->owner->cid.cid_tns = chan->cid.cid_tns;
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channel->owner->adsicpe = chan->adsicpe;
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channel->owner->transfercapability = chan->transfercapability;
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}
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/* Actually call the device */
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if ((res = ast_call(channel->owner, numsubst, 0))) {
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@ -530,12 +534,16 @@ enum ast_dial_result ast_dial_run(struct ast_dial *dial, struct ast_channel *cha
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enum ast_dial_result res = AST_DIAL_RESULT_TRYING;
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/* Ensure required arguments are passed */
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if (!dial || !chan)
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if (!dial || (!chan && !async)) {
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ast_log(LOG_DEBUG, "invalid #1\n");
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return AST_DIAL_RESULT_INVALID;
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}
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/* If there are no channels to dial we can't very well try to dial them */
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if (AST_LIST_EMPTY(&dial->channels))
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if (AST_LIST_EMPTY(&dial->channels)) {
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ast_log(LOG_DEBUG, "invalid #2\n");
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return AST_DIAL_RESULT_INVALID;
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}
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/* Dial each requested channel */
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if (!begin_dial(dial, chan))
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@ -543,6 +551,7 @@ enum ast_dial_result ast_dial_run(struct ast_dial *dial, struct ast_channel *cha
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/* If we are running async spawn a thread and send it away... otherwise block here */
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if (async) {
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dial->status = AST_DIAL_RESULT_TRYING;
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/* Try to create a thread */
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if (ast_pthread_create(&dial->thread, NULL, async_dial, dial)) {
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/* Failed to create the thread - hangup all dialed channels and return failed */
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