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Diffstat (limited to 'trunk/configs/h323.conf.sample')
| -rw-r--r-- | trunk/configs/h323.conf.sample | 207 |
1 files changed, 0 insertions, 207 deletions
diff --git a/trunk/configs/h323.conf.sample b/trunk/configs/h323.conf.sample deleted file mode 100644 index 5be321f33..000000000 --- a/trunk/configs/h323.conf.sample +++ /dev/null @@ -1,207 +0,0 @@ -; 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 qos.tex or Quality of Service section of asterisk.pdf 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 doc/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 - - - |