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 <[EMAIL PROTECTED]>
;
; 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
;
; 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
; [EMAIL PROTECTED]
; secret=sj25pwd
;
; 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://[EMAIL PROTECTED]
;exten => 725,1,Playback,transfer|skip
;exten => 725,1,Dial,SIP/sj25
