[asterisk-commits] mattf: trunk r43288 -
/trunk/configs/h323.conf.sample
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Sep 19 12:25:18 MST 2006
Author: mattf
Date: Tue Sep 19 14:25:18 2006
New Revision: 43288
URL: http://svn.digium.com/view/asterisk?rev=43288&view=rev
Log:
Add the h323 config file. Arrr!!! for international talk like a pirate's day.
Added:
trunk/configs/h323.conf.sample (with props)
Added: trunk/configs/h323.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/h323.conf.sample?rev=43288&view=auto
==============================================================================
--- trunk/configs/h323.conf.sample (added)
+++ trunk/configs/h323.conf.sample Tue Sep 19 14:25:18 2006
@@ -1,0 +1,192 @@
+; 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
+;tos=lowdelay
+;
+; 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 :)
+;
+; User-Input Mode (DTMF)
+;
+; valid entries are: rfc2833, inband
+; 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.
+;
+; You may also specify on either a per-peer or per-user basis below.
+;dtmfcodec=101
+;
+; 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
+;
+;------------------------------ 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 at 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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