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- ; 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
- ;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
- ; preferences. Defaults to no.
- ;
- ; 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
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