[Asterisk-Users] calls between SIP and H.323 clients

[Truong]

Hello all,

It's me again.

I would like play with calls between a H.323 client and a SIP client
through * inside my LAN.

For that,
on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk;
on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I
register into * with a username, no password. The 3 files oh323.conf,
sip.conf, extensions.conf are in attachment.

In the same context [voip], I defined an extension "665" for
OH323/192.168.1.25 (not used in pratique), and "725" for SIP/sj25.

I place, from GM20, a callto://192.168.1.20 and then the extension "725"
for being be routed into SJ25. It works ! I was happy with the result.

I need helps to do these:

 1. separate the actual context [voip] into two: [h323] and [sip]. And
 it could be nice if I have a third context like [voip] to invite caller
 to choose [h323] or [sip], etc.

 2. if a caller knows an extension in [h323] or [sip], he could place a
 call to it directly without going through the voice menu of [voip]


Thank you very much for your hints.

-- 
  Truong <tphuong at wol.be>
-------------- next part --------------
;
; Configuration file of OpenH323 channel driver
;

;-----------------------------------------
; General configuration options
; (ports, jitter, GK, ...)
;-----------------------------------------
[general]
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720
fastStart=yes
h245Tunnelling=yes
h245inSetup=yes
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
libTraceLevel=1
libTraceFile=stdout

;gatekeeper=192.168.1.2
;gatekeeper=DISCOVER
;gatekeeperPassword=secret
gatekeeper=DISABLE
;
; Set the mode for sending user-input
; Valid values for this option are:
;	Q931		-	Q.931 Keypad Information Element
;	STRING		-	H.245 string
;	TONE		-	H.245 tone
;	RFC2833		-	RFC2833
;
userInputMode=TONE
amaFlags=default
accountCode=H323

context=voip

;-----------------------------------------
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-----------------------------------------
[register]
alias=asterisk
alias=123
context=all-aliases
alias=ASTERISK
alias=666

context=voip
alias=665

;-----------------------------------------
; Specify and configure CODEC related
; options
;-----------------------------------------
[codecs]
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
; Valid values for the "codec" option are:
;	G711U		-	G.711 u-Law
;	G711A		-	G.711 A-Law
;	G7231		-	G.723.1(6.3k)
;	G72316K3	-	G.723.1(6.3k)
;	G72315K3	-	G.723.1(5.3k)
;	G7231A6K3	-	G.723.1A(6.3k)
;	G7231A6K3	-	G.723.1A(6.3k)
;	G728		-	G.728
;	G729		-	G.729
;	G729A		-	G.729A
;	G729B		-	G.729B
;	G729AB		-	G.729AB
;	GSM0610		-	GSM 0610
;	MSGSM		-	Microsoft GSM Audio Capability
;	LPC10		-	LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;	
codec=GSM0610
frames=4
codec=G711A
frames=20
;codec=G7231

-------------- next part --------------
;
; SIP Configuration for Asterisk
;
[general]
port = 5060			; Port to bind to
bindaddr = 0.0.0.0		; Address to bind to
context = default		; Default for incoming calls
; allow=ulaw

[sj25]
type=friend
dtmfmode=inband		        ; Choices are inband, rfc2833, or info
host=dynamic
context = voip
;
; defaultip=192.168.1.25
; mailbox=1234,2345		; Mailbox for message waiting indicator
; username=sj25 at 192.168.1.25
; secret=sj25pwd

-------------- next part --------------
;
; Static extension configuration files, used by
; the pbx_config module.
;
; The "General" category is for certain variables.  
;
[general]
static=yes
writeprotect=no

;
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
; variable
;
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
;CONSOLE=Console/dsp				; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
;IAXINFO=guest					; IAXtel username/password

[start]
;
; from [demo]
exten => s,1,Wait,1			; Wait a second, just for fun
exten => s,2,Answer			; Answer the line
exten => s,3,DigitTimeout,5		; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10		; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats)	; Play a congratulatory message

; ok pour chan_oh323.so et oh323.conf
[voip]
;
include => start
;
;****************
; common option for all extensions
;  s = start ; t = time-out ; i = invalid
;****************
;
; It works for direct incoming call with the line below:
;exten => s,1,Dial,OH323/192.168.1.25
;

;-- H.323 [alias = 665]
exten => 665,1,Dial(OH323/192.168.1.25)
exten => 725,1,Dial,SIP/sj25


; this context [sip] is defined in sip.conf
;[sip]
;
;include => start
;
;-- SIP [sip://725@192.168.1.25]
;exten => 725,1,Playback,transfer|skip
;exten => 725,1,Dial,SIP/sj25