Remove files left behind on removal of h323, jingle and jabber.
This change removes h323.conf.sample, jingle.h, jabber.h left behind by r3698. Review: https://reviewboard.asterisk.org/r/3755/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Corey Farrell
parent
0d1288e2d2
commit
6461d90d8a
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@ -1,210 +0,0 @@
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; The NuFone Network's
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; Open H.323 driver configuration
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;
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[general]
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port = 1720
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;bindaddr = 1.2.3.4 ; this SHALL contain a single, valid IP address for this machine
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;
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; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
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;
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; You may specify a global default AMA flag for iaxtel calls. It must be
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; one of 'default', 'omit', 'billing', or 'documentation'. These flags
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; are used in the generation of call detail records.
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;
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;amaflags = default
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;
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; You may specify a default account for Call Detail Records in addition
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; to specifying on a per-user basis
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;
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;accountcode=lss0101
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;
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; You can fine tune codecs here using "allow" and "disallow" clauses
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; with specific codecs. Use "all" to represent all formats.
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;
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;disallow=all
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;allow=all ; turns on all installed codecs
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;disallow=g723.1 ; Hm... Proprietary, don't use it...
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;allow=gsm ; Always allow GSM, it's cool :)
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;allow=ulaw ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
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; for framing options
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;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
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; preferences. Defaults to no.
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;
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; User-Input Mode (DTMF)
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;
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; valid entries are: rfc2833, inband, cisco, h245-signal
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; default is rfc2833
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;dtmfmode=rfc2833
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;
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; Default RTP Payload to send RFC2833 DTMF on. This is used to
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; interoperate with broken gateways which cannot successfully
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; negotiate a RFC2833 payload type in the TerminalCapabilitySet.
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; To specify required payload type, put it after colon in dtmfmode
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; option like
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;dtmfmode=rfc2833:101
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; or
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;dtmfmode=cisco:121
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;
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; Set the gatekeeper
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; DISCOVER - Find the Gk address using multicast
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; DISABLE - Disable the use of a GK
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; <IP address> or <Host name> - The acutal IP address or hostname of your GK
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;gatekeeper = DISABLE
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;
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;
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; Tell Asterisk whether or not to accept Gatekeeper
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; routed calls or not. Normally this should always
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; be set to yes, unless you want to have finer control
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; over which users are allowed access to Asterisk.
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; Default: YES
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;
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;AllowGKRouted = yes
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;
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; When the channel works without gatekeeper, there is possible to
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; reject calls from anonymous (not listed in users) callers.
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; Default is to allow anonymous calls.
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;
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;AcceptAnonymous = yes
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;
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; Optionally you can determine a user by Source IP versus its H.323 alias.
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; Default behavour is to determine user by H.323 alias.
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;
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;UserByAlias=no
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;
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; Default context gets used in siutations where you are using
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; the GK routed model or no type=user was found. This gives you
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; the ability to either play an invalid message or to simply not
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; use user authentication at all.
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;
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;context=default
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;
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; Use this option to help Cisco (or other) gateways to setup backward voice
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; path to pass inband tones to calling user (see, for example,
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; http://www.cisco.com/warp/public/788/voip/ringback.html)
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;
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; Add PROGRESS information element to SETUP message sent on outbound calls
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; to notify about required backward voice path. Valid values are:
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; 0 - don't add PROGRESS information element (default);
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; 1 - call is not end-end ISDN, further call progress information can
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; possibly be available in-band;
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; 3 - origination address is non-ISDN (Cisco accepts this value only);
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; 8 - in-band information or an appropriate pattern is now available;
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;progress_setup = 3
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;
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; Add PROGRESS information element (IE) to ALERT message sent on incoming
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; calls to notify about required backwared voice path. Valid values are:
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; 0 - don't add PROGRESS IE (default);
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; 8 - in-band information or an appropriate pattern is now available;
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;progress_alert = 8
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;
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; Generate PROGRESS message when H.323 audio path has established to create
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; backward audio path at other end of a call.
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;progress_audio = yes
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;
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; Specify how to inject non-standard information into H.323 messages. When
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; the channel receives messages with tunneled information, it automatically
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; enables the same option for all further outgoing messages independedly on
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; options has been set by the configuration. This behavior is required, for
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; example, for Cisco CallManager when Q.SIG tunneling is enabled for a
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; gateway where Asterisk lives.
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; The option can be used multiple times, one option per line.
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;tunneling=none ; Totally disable tunneling (default)
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;tunneling=cisco ; Enable Cisco-specific tunneling
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;tunneling=qsig ; Enable tunneling via Q.SIG messages
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;
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; Specify how to pass hold notification to remote party. Default is to
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; use H.450.4 supplementary service message.
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;hold=none ; Do not pass hold/retrieve notifications
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;hold=notify ; Use H.225 NOTIFY message
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;hold=q931only ; Use stripped H.225 NOTIFY message (Q.931 part
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; ; only, usable for Cisco CallManager)
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;hold=h450 ; Pass notification as H.450.4 supplementary
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; ; service
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;
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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; H323 channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The H323 channel can accept jitter,
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; thus an enabled jitterbuffer on the receive H323 side will only
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; be used if the sending side can create jitter and jbforce is
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; also set to yes.
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; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a H323
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; channel. Defaults to "no".
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; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usualy sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323
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; channel. Two implementations are currenlty available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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;
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; H.323 Alias definitions
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;
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; Type 'h323' will register aliases to the endpoint
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; and Gatekeeper, if there is one.
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;
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; Example: if someone calls time@your.asterisk.box.com
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; Asterisk will send the call to the extension 'time'
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; in the context default
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;
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; [default]
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; exten => time,1,Answer
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; exten => time,2,Playback,current-time
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;
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; Keyword's 'prefix' and 'e164' are only make sense when
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; used with a gatekeeper. You can specify either a prefix
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; or E.164 this endpoint is responsible for terminating.
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;
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; Example: The H.323 alias 'det-gw' will tell the gatekeeper
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; to route any call with the prefix 1248 to this alias. Keyword
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; e164 is used when you want to specifiy a full telephone
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; number. So a call to the number 18102341212 would be
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; routed to the H.323 alias 'time'.
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;
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;[time]
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;type=h323
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;e164=18102341212
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;context=default
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;
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;[det-gw]
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;type=h323
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;prefix=1248,1313
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;context=detroit
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;
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;
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; Inbound H.323 calls from BillyBob would land in the incoming
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; context with a maximum of 4 concurrent incoming calls
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;
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;
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; Note: If keyword 'incominglimit' are omitted Asterisk will not
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; enforce any maximum number of concurrent calls.
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;
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;[BillyBob]
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;type=user
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;host=192.168.1.1
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;context=incoming
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;incominglimit=4
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;h245Tunneling=no
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;
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;
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; Outbound H.323 call to Larry using SlowStart
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;
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;[Larry]
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;type=peer
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;host=192.168.2.1
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;fastStart=no
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@ -1,224 +0,0 @@
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2010, Digium, Inc.
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*
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* Matt O'Gorman <mogorman@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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* \brief AJI - The Asterisk Jabber Interface
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* \arg \ref AJI_intro
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* \ref res_jabber.c
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* \author Matt O'Gorman <mogorman@digium.com>
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* IKSEMEL http://iksemel.jabberstudio.org
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*
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* \page AJI_intro AJI - The Asterisk Jabber Interface
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*
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* The Asterisk Jabber Interface, AJI, publishes an API for
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* modules to use jabber communication. res_jabber.c implements
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* a Jabber client and a component that can connect as a service
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* to Jabber servers.
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*
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* \section External dependencies
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* AJI use the IKSEMEL library found at http://iksemel.jabberstudio.org/
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*
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* \section Files
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* - res_jabber.c
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* - jabber.h
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* - chan_gtalk.c
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*
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*/
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#ifndef _ASTERISK_JABBER_H
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#define _ASTERISK_JABBER_H
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#ifdef HAVE_OPENSSL
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#include <openssl/ssl.h>
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#include <openssl/err.h>
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#define TRY_SECURE 2
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#define SECURE 4
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#endif /* HAVE_OPENSSL */
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/* file is read by blocks with this size */
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#define NET_IO_BUF_SIZE 4096
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/* Return value for timeout connection expiration */
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#define IKS_NET_EXPIRED 12
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#include <iksemel.h>
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#include "asterisk/astobj.h"
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#include "asterisk/linkedlists.h"
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/*
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* As per RFC 3920 - section 3.1, the maximum length for a full Jabber ID
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* is 3071 bytes.
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* The ABNF syntax for jid :
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* jid = [node "@" ] domain [ "/" resource ]
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* Each allowable portion of a JID (node identifier, domain identifier,
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* and resource identifier) MUST NOT be more than 1023 bytes in length,
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* resulting in a maximum total size (including the '@' and '/' separators)
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* of 3071 bytes.
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*/
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#define AJI_MAX_JIDLEN 3071
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#define AJI_MAX_RESJIDLEN 1023
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#define AJI_MAX_ATTRLEN 256
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#define MUC_NS "http://jabber.org/protocol/muc"
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enum aji_state {
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AJI_DISCONNECTING,
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AJI_DISCONNECTED,
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AJI_CONNECTING,
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AJI_CONNECTED
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};
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enum {
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AJI_AUTOPRUNE = (1 << 0),
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AJI_AUTOREGISTER = (1 << 1),
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AJI_AUTOACCEPT = (1 << 2),
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};
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enum {
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AJI_XEP0248 = (1 << 0),
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AJI_PUBSUB = (1 << 1),
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AJI_PUBSUB_AUTOCREATE = (1 << 2),
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};
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enum aji_btype {
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AJI_USER = 0,
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AJI_TRANS = 1,
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AJI_UTRANS = 2,
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};
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struct aji_version {
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char version[50];
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int jingle;
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struct aji_capabilities *parent;
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struct aji_version *next;
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};
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struct aji_capabilities {
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char node[200];
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struct aji_version *versions;
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struct aji_capabilities *next;
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};
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struct aji_resource {
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int status;
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char resource[AJI_MAX_RESJIDLEN];
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char *description;
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struct aji_version *cap;
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int priority;
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struct aji_resource *next;
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};
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struct aji_message {
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char *from;
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char *message;
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char id[25];
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struct timeval arrived;
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AST_LIST_ENTRY(aji_message) list;
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};
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struct aji_buddy {
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ASTOBJ_COMPONENTS_FULL(struct aji_buddy, AJI_MAX_JIDLEN, 1);
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char channel[160];
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struct aji_resource *resources;
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enum aji_btype btype;
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struct ast_flags flags;
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};
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struct aji_buddy_container {
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ASTOBJ_CONTAINER_COMPONENTS(struct aji_buddy);
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};
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struct aji_transport_container {
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ASTOBJ_CONTAINER_COMPONENTS(struct aji_transport);
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};
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struct aji_client {
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ASTOBJ_COMPONENTS(struct aji_client);
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char password[160];
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char user[AJI_MAX_JIDLEN];
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char serverhost[AJI_MAX_RESJIDLEN];
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char pubsub_node[AJI_MAX_RESJIDLEN];
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char statusmessage[256];
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char name_space[256];
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char sid[10]; /* Session ID */
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char mid[6]; /* Message ID */
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char context[AST_MAX_CONTEXT];
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iksid *jid;
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iksparser *p;
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iksfilter *f;
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ikstack *stack;
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#ifdef HAVE_OPENSSL
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SSL_CTX *ssl_context;
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SSL *ssl_session;
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const SSL_METHOD *ssl_method;
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unsigned int stream_flags;
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#endif /* HAVE_OPENSSL */
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enum aji_state state;
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int port;
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int debug;
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int usetls;
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int forcessl;
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int usesasl;
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int keepalive;
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int allowguest;
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int timeout;
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int message_timeout;
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int authorized;
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int distribute_events;
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int send_to_dialplan;
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struct ast_flags flags;
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int component; /* 0 client, 1 component */
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struct aji_buddy_container buddies;
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AST_LIST_HEAD(messages,aji_message) messages;
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void *jingle;
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pthread_t thread;
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int priority;
|
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enum ikshowtype status;
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};
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struct aji_client_container{
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ASTOBJ_CONTAINER_COMPONENTS(struct aji_client);
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};
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/* !Send XML stanza over the established XMPP connection */
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int ast_aji_send(struct aji_client *client, iks *x);
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/*! Send jabber chat message from connected client to jabber URI */
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int ast_aji_send_chat(struct aji_client *client, const char *address, const char *message);
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/*! Send jabber chat message from connected client to a groupchat using
|
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* a given nickname */
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int ast_aji_send_groupchat(struct aji_client *client, const char *nick, const char *address, const char *message);
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/*! Disconnect jabber client */
|
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int ast_aji_disconnect(struct aji_client *client);
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int ast_aji_check_roster(void);
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void ast_aji_increment_mid(char *mid);
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/*! Open Chat session */
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int ast_aji_create_chat(struct aji_client *client,char *room, char *server, char *topic);
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/*! Invite to opened Chat session */
|
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int ast_aji_invite_chat(struct aji_client *client, char *user, char *room, char *message);
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/*! Join/leave existing Chat session */
|
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int ast_aji_join_chat(struct aji_client *client, char *room, char *nick);
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int ast_aji_leave_chat(struct aji_client *client, char *room, char *nick);
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/*! Get a client via its name. Increases refcount of client by 1 */
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struct aji_client *ast_aji_get_client(const char *name);
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struct aji_client_container *ast_aji_get_clients(void);
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/*! Destructor function for buddies to be used with ASTOBJ_UNREF */
|
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void ast_aji_buddy_destroy(struct aji_buddy *obj);
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/*! Destructor function for clients to be used with ASTOBJ_UNREF after calls to ast_aji_get_client */
|
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void ast_aji_client_destroy(struct aji_client *obj);
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#endif
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@ -1,66 +0,0 @@
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/*
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* Asterisk -- An open source telephony toolkit.
|
||||
*
|
||||
* Copyright (C) 1999 - 2005, Digium, Inc.
|
||||
*
|
||||
* Matt O'Gorman <mogorman@digium.com>
|
||||
*
|
||||
* See http://www.asterisk.org for more information about
|
||||
* the Asterisk project. Please do not directly contact
|
||||
* any of the maintainers of this project for assistance;
|
||||
* the project provides a web site, mailing lists and IRC
|
||||
* channels for your use.
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License Version 2. See the LICENSE file
|
||||
* at the top of the source tree.
|
||||
*/
|
||||
|
||||
/*! \file
|
||||
* \brief Jingle definitions for chan_jingle
|
||||
*
|
||||
* \ref chan_jingle.c
|
||||
*
|
||||
* \author Matt O'Gorman <mogorman@digium.com>
|
||||
*/
|
||||
|
||||
|
||||
#ifndef _ASTERISK_JINGLE_H
|
||||
#define _ASTERISK_JINGLE_H
|
||||
|
||||
#include <iksemel.h>
|
||||
#include "asterisk/astobj.h"
|
||||
|
||||
|
||||
/* Jingle Constants */
|
||||
|
||||
#define JINGLE_NODE "jingle"
|
||||
#define GOOGLE_NODE "session"
|
||||
|
||||
#define JINGLE_NS "urn:xmpp:tmp:jingle"
|
||||
#define JINGLE_AUDIO_RTP_NS "urn:xmpp:tmp:jingle:apps:audio-rtp"
|
||||
#define JINGLE_VIDEO_RTP_NS "urn:xmpp:tmp:jingle:apps:video"
|
||||
#define JINGLE_ICE_UDP_NS "urn:xmpp:tmp:jingle:transports:ice-udp"
|
||||
#define JINGLE_DTMF_NS "urn:xmpp:tmp:jingle:dtmf"
|
||||
|
||||
#define GOOGLE_NS "http://www.google.com/session"
|
||||
#define GOOGLE_JINGLE_NS "urn:xmpp:jingle:1"
|
||||
#define GOOGLE_AUDIO_NS "http://www.google.com/session/phone"
|
||||
#define GOOGLE_VIDEO_NS "http://www.google.com/session/video"
|
||||
#define GOOGLE_TRANSPORT_NS "http://www.google.com/transport/p2p"
|
||||
|
||||
#define JINGLE_SID "sid"
|
||||
#define GOOGLE_SID "id"
|
||||
|
||||
#define JINGLE_INITIATE "session-initiate"
|
||||
|
||||
#define JINGLE_ACCEPT "session-accept"
|
||||
#define GOOGLE_ACCEPT "accept"
|
||||
|
||||
#define JINGLE_NEGOTIATE "transport-info"
|
||||
#define GOOGLE_NEGOTIATE "candidates"
|
||||
|
||||
#define JINGLE_INFO "session-info"
|
||||
#define JINGLE_TERMINATE "session-terminate"
|
||||
|
||||
#endif
|
||||
Loading…
Reference in New Issue