mirror of https://github.com/asterisk/asterisk
You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
209 lines
7.9 KiB
209 lines
7.9 KiB
; The NuFone Network's
|
|
; Open H.323 driver configuration
|
|
;
|
|
[general]
|
|
port = 1720
|
|
;bindaddr = 1.2.3.4 ; this SHALL contain a single, valid IP address for this machine
|
|
;
|
|
; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
|
|
;tos_audio=ef ; Sets TOS for RTP audio packets.
|
|
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
|
|
;
|
|
; You may specify a global default AMA flag for iaxtel calls. It must be
|
|
; one of 'default', 'omit', 'billing', or 'documentation'. These flags
|
|
; are used in the generation of call detail records.
|
|
;
|
|
;amaflags = default
|
|
;
|
|
; You may specify a default account for Call Detail Records in addition
|
|
; to specifying on a per-user basis
|
|
;
|
|
;accountcode=lss0101
|
|
;
|
|
; You can fine tune codecs here using "allow" and "disallow" clauses
|
|
; with specific codecs. Use "all" to represent all formats.
|
|
;
|
|
;disallow=all
|
|
;allow=all ; turns on all installed codecs
|
|
;disallow=g723.1 ; Hm... Proprietary, don't use it...
|
|
;allow=gsm ; Always allow GSM, it's cool :)
|
|
;allow=ulaw ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
|
|
; for framing options
|
|
;
|
|
; User-Input Mode (DTMF)
|
|
;
|
|
; valid entries are: rfc2833, inband, cisco, h245-signal
|
|
; default is rfc2833
|
|
;dtmfmode=rfc2833
|
|
;
|
|
; Default RTP Payload to send RFC2833 DTMF on. This is used to
|
|
; interoperate with broken gateways which cannot successfully
|
|
; negotiate a RFC2833 payload type in the TerminalCapabilitySet.
|
|
; To specify required payload type, put it after colon in dtmfmode
|
|
; option like
|
|
;dtmfmode=rfc2833:101
|
|
; or
|
|
;dtmfmode=cisco:121
|
|
;
|
|
; Set the gatekeeper
|
|
; DISCOVER - Find the Gk address using multicast
|
|
; DISABLE - Disable the use of a GK
|
|
; <IP address> or <Host name> - The acutal IP address or hostname of your GK
|
|
;gatekeeper = DISABLE
|
|
;
|
|
;
|
|
; Tell Asterisk whether or not to accept Gatekeeper
|
|
; routed calls or not. Normally this should always
|
|
; be set to yes, unless you want to have finer control
|
|
; over which users are allowed access to Asterisk.
|
|
; Default: YES
|
|
;
|
|
;AllowGKRouted = yes
|
|
;
|
|
; When the channel works without gatekeeper, there is possible to
|
|
; reject calls from anonymous (not listed in users) callers.
|
|
; Default is to allow anonymous calls.
|
|
;
|
|
;AcceptAnonymous = yes
|
|
;
|
|
; Optionally you can determine a user by Source IP versus its H.323 alias.
|
|
; Default behavour is to determine user by H.323 alias.
|
|
;
|
|
;UserByAlias=no
|
|
;
|
|
; Default context gets used in siutations where you are using
|
|
; the GK routed model or no type=user was found. This gives you
|
|
; the ability to either play an invalid message or to simply not
|
|
; use user authentication at all.
|
|
;
|
|
;context=default
|
|
;
|
|
; Use this option to help Cisco (or other) gateways to setup backward voice
|
|
; path to pass inband tones to calling user (see, for example,
|
|
; http://www.cisco.com/warp/public/788/voip/ringback.html)
|
|
;
|
|
; Add PROGRESS information element to SETUP message sent on outbound calls
|
|
; to notify about required backward voice path. Valid values are:
|
|
; 0 - don't add PROGRESS information element (default);
|
|
; 1 - call is not end-end ISDN, further call progress information can
|
|
; possibly be available in-band;
|
|
; 3 - origination address is non-ISDN (Cisco accepts this value only);
|
|
; 8 - in-band information or an appropriate pattern is now available;
|
|
;progress_setup = 3
|
|
;
|
|
; Add PROGRESS information element (IE) to ALERT message sent on incoming
|
|
; calls to notify about required backwared voice path. Valid values are:
|
|
; 0 - don't add PROGRESS IE (default);
|
|
; 8 - in-band information or an appropriate pattern is now available;
|
|
;progress_alert = 8
|
|
;
|
|
; Generate PROGRESS message when H.323 audio path has established to create
|
|
; backward audio path at other end of a call.
|
|
;progress_audio = yes
|
|
;
|
|
; Specify how to inject non-standard information into H.323 messages. When
|
|
; the channel receives messages with tunneled information, it automatically
|
|
; enables the same option for all further outgoing messages independedly on
|
|
; options has been set by the configuration. This behavior is required, for
|
|
; example, for Cisco CallManager when Q.SIG tunneling is enabled for a
|
|
; gateway where Asterisk lives.
|
|
; The option can be used multiple times, one option per line.
|
|
;tunneling=none ; Totally disable tunneling (default)
|
|
;tunneling=cisco ; Enable Cisco-specific tunneling
|
|
;tunneling=qsig ; Enable tunneling via Q.SIG messages
|
|
;
|
|
; Specify how to pass hold notification to remote party. Default is to
|
|
; use H.450.4 supplementary service message.
|
|
;hold=none ; Do not pass hold/retrieve notifications
|
|
;hold=notify ; Use H.225 NOTIFY message
|
|
;hold=q931only ; Use stripped H.225 NOTIFY message (Q.931 part
|
|
; ; only, usable for Cisco CallManager)
|
|
;hold=h450 ; Pass notification as H.450.4 supplementary
|
|
; ; service
|
|
;
|
|
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
|
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
|
; H323 channel. Defaults to "no". An enabled jitterbuffer will
|
|
; be used only if the sending side can create and the receiving
|
|
; side can not accept jitter. The H323 channel can accept jitter,
|
|
; thus an enabled jitterbuffer on the receive H323 side will only
|
|
; be used if the sending side can create jitter and jbforce is
|
|
; also set to yes.
|
|
|
|
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a H323
|
|
; channel. Defaults to "no".
|
|
|
|
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
|
|
|
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
|
; resynchronized. Useful to improve the quality of the voice, with
|
|
; big jumps in/broken timestamps, usualy sent from exotic devices
|
|
; and programs. Defaults to 1000.
|
|
|
|
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323
|
|
; channel. Two implementations are currenlty available - "fixed"
|
|
; (with size always equals to jbmax-size) and "adaptive" (with
|
|
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
|
|
|
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
|
;-----------------------------------------------------------------------------------
|
|
;
|
|
; H.323 Alias definitions
|
|
;
|
|
; Type 'h323' will register aliases to the endpoint
|
|
; and Gatekeeper, if there is one.
|
|
;
|
|
; Example: if someone calls time@your.asterisk.box.com
|
|
; Asterisk will send the call to the extension 'time'
|
|
; in the context default
|
|
;
|
|
; [default]
|
|
; exten => time,1,Answer
|
|
; exten => time,2,Playback,current-time
|
|
;
|
|
; Keyword's 'prefix' and 'e164' are only make sense when
|
|
; used with a gatekeeper. You can specify either a prefix
|
|
; or E.164 this endpoint is responsible for terminating.
|
|
;
|
|
; Example: The H.323 alias 'det-gw' will tell the gatekeeper
|
|
; to route any call with the prefix 1248 to this alias. Keyword
|
|
; e164 is used when you want to specifiy a full telephone
|
|
; number. So a call to the number 18102341212 would be
|
|
; routed to the H.323 alias 'time'.
|
|
;
|
|
;[time]
|
|
;type=h323
|
|
;e164=18102341212
|
|
;context=default
|
|
;
|
|
;[det-gw]
|
|
;type=h323
|
|
;prefix=1248,1313
|
|
;context=detroit
|
|
;
|
|
;
|
|
; Inbound H.323 calls from BillyBob would land in the incoming
|
|
; context with a maximum of 4 concurrent incoming calls
|
|
;
|
|
;
|
|
; Note: If keyword 'incominglimit' are omitted Asterisk will not
|
|
; enforce any maximum number of concurrent calls.
|
|
;
|
|
;[BillyBob]
|
|
;type=user
|
|
;host=192.168.1.1
|
|
;context=incoming
|
|
;incominglimit=4
|
|
;h245Tunneling=no
|
|
;
|
|
;
|
|
; Outbound H.323 call to Larry using SlowStart
|
|
;
|
|
;[Larry]
|
|
;type=peer
|
|
;host=192.168.2.1
|
|
;fastStart=no
|
|
|
|
|
|
|