asterisk/configs/h323.conf.sample

; 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