exten => 23,1,NoOp(${STRFTIME(${EPOCH},,%d%m%H:%M)})
exten => 23,2,Playback(num:${STRFTIME(${EPOCH},,%H)}|say);
exten => 23,n,Playback(digits/oclock);
exten => 23,n,Playback(num:${STRFTIME(${EPOCH},,%M)}|say);
exten => 23,n,Playback(digits/minutes);
exten => 23,n,Playback(num:${STRFTIME(${EPOCH},,%S)}|say);
exten => 23,n,Playback(digits/seconds);
exten => 23,n,Hangup()

debian:/etc/asterisk# cat h323.conf
; 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 :)
allow=ulaw ; see doc/rtp-packetization for framing options

allow=alaw ; see doc/rtp-packetization for framing options
allow=g729 ; see doc/rtp-packetization for framing options

;
; 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=local_users
;
; 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=n


[5000]
type=friend
;host=192.168.1.1
faststart=yes
context=from_local_users
;incominglimit=4
h245Tunneling=yes
;

*CLI>
*CLI> == New H.323 Connection created.
--Received SETUP message
-- Setting up Call
-- Call token: [ip$172.16.100.44:14031/26062]
-- Calling party name: [voip.172.16.100.44]
-- Calling party number: [5000]
-- Called party name: [23]
-- Called party number: [23]
-- Calling party IP: [172.16.100.44]
Setting capabilities to 0x10e (gsm|ulaw|alaw|g729)
Capabilities in preference order is (gsm|ulaw|alaw|g729)
Allowed Codecs:
Table:
GSM-06.10 <1>
G.711-uLaw-64k <2>
G.711-ALaw-64k <3>
G.729A <4>
G.729 <5>
UserInput/hookflash <6>
UserInput/RFC2833 <7>
UserInput/dtmf <8>
Set:
0:
0:
GSM-06.10 <1>
G.711-uLaw-64k <2>
G.711-ALaw-64k <3>
G.729A <4>
G.729 <5>
1:
UserInput/hookflash <6>
2:
UserInput/RFC2833 <7>
UserInput/dtmf <8>

=-= In OnAnswerCall for call 26062
- Progress Indicator: 0
- Inserting PI of 0 into ALERTING message
-- Started logical channel: sending G.729
-- channelsOpen = 1
External RTP Session Starting
RTP channel id 1 parameters:
-- remoteIpAddress: 172.16.100.44
-- remotePort: 23062
-- ExternalIpAddress: 127.0.0.1
-- ExternalPort: 14958
-- Started logical channel: receiving G.729
-- channelsOpen = 2
External RTP Session Starting
RTP channel id 1 parameters:
-- remoteIpAddress: 172.16.100.44
-- remotePort: 23062
-- ExternalIpAddress: 127.0.0.1
-- ExternalPort: 14958
-- Executing [23@from_local_users:1] NoOp("H323/ip$172.16.100.44:14031/26062", "100716:41") in new stack
-- Executing [23@from_local_users:2] Playback("H323/ip$172.16.100.44:14031/26062", "num:16|say") in new stack
Answering call ip$172.16.100.44:14031/26062
-- <H323/ip$172.16.100.44:14031/26062> Playing 'digits/16' (language 'en')
[Jul 10 16:41:44] WARNING[18403]: file.c:1198 waitstream_core: Unexpected control subclass '14'
[Jul 10 16:41:44] WARNING[18403]: file.c:1198 waitstream_core: Unexpected control subclass '14'
-- Received Facility message...
-- Inbound RFC2833 on payload [pt=101]
Peer capability is G.729 <1>
Found peer capability G.729 <1>, Asterisk code is 256, frame size (in ms) is 20
Peer capability is UserInput/basicString <3>
Peer capability is UserInput/hookflash <4>
Peer capability is UserInput/RFC2833 <5>
Peer capabilities = 0x100 (g729), ordered list is (g729)
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
=-= In OnConnectionEstablished for call 26062
-- Connection Established with "voip.172.16.100.44 (5000) [172.16.100.44]"
-- Executing [23@from_local_users:3] Playback("H323/ip$172.16.100.44:14031/26062", "digits/oclock") in new stack
-- <H323/ip$172.16.100.44:14031/26062> Playing 'digits/oclock' (language 'en')
-- Executing [23@from_local_users:4] Playback("H323/ip$172.16.100.44:14031/26062", "num:41|say") in new stack
-- <H323/ip$172.16.100.44:14031/26062> Playing 'digits/40' (language 'en')
-- <H323/ip$172.16.100.44:14031/26062> Playing 'digits/1' (language 'en')
-- Executing [23@from_local_users:5] Playback("H323/ip$172.16.100.44:14031/26062", "digits/minutes") in new stack
-- <H323/ip$172.16.100.44:14031/26062> Playing 'digits/minutes' (language 'en')
-- Executing [23@from_local_users:6] Playback("H323/ip$172.16.100.44:14031/26062", "num:48|say") in new stack
-- <H323/ip$172.16.100.44:14031/26062> Playing 'digits/40' (language 'en')
-- <H323/ip$172.16.100.44:14031/26062> Playing 'digits/8' (language 'en')
-- Executing [23@from_local_users:7] Playback("H323/ip$172.16.100.44:14031/26062", "digits/seconds") in new stack
-- <H323/ip$172.16.100.44:14031/26062> Playing 'digits/seconds' (language 'en')
-- Executing [23@from_local_users:8] Hangup("H323/ip$172.16.100.44:14031/26062", "") in new stack
== Spawn extension (from_local_users, 23, 8) exited non-zero on 'H323/ip$172.16.100.44:14031/26062'
-- Sending RELEASE COMPLETE
-- ClearCall: Request to clear call with token ip$172.16.100.44:14031/26062, cause EndedByRemoteUser
channelsOpen = 1
channelsOpen = 0
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
-- ClearCall: Request to clear call with token ip$172.16.100.44:14031/26062, cause EndedByTransportFail
-- voip.172.16.100.44 (5000) [172.16.100.44] has cleared the call
== H.323 Connection deleted.