From 2e5118cc49b551ed9da5b62cacbd99d734932760 Mon Sep 17 00:00:00 2001 From: Matthew Fredrickson Date: Tue, 19 Sep 2006 19:25:18 +0000 Subject: Add the h323 config file. Arrr!!! for international talk like a pirate's day. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43288 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/h323.conf.sample | 192 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 192 insertions(+) create mode 100644 configs/h323.conf.sample (limited to 'configs') diff --git a/configs/h323.conf.sample b/configs/h323.conf.sample new file mode 100644 index 000000000..1962f2167 --- /dev/null +++ b/configs/h323.conf.sample @@ -0,0 +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 +; or - 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@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 + + + -- cgit v1.2.3