211 lines
8.0 KiB
Plaintext
211 lines
8.0 KiB
Plaintext
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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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