[Asterisk-Users] SIP Model and H323

[Michael Manousos]

Carlos Crembil wrote:
> Hi guys,
> I'm new here, so, greatings for all... (i'll give you the candies in a
> future meeting :-).
> 
> I've installed asterisk and opengk in my server, and I'm in the
> experimenting phase. Also I have a Cisco 800 series to play (4 FXS
> interfaces), and a netmeeting client.
> 
> My actual configuration is H323 based. My Cisco can call asterisk, and my
> netmeeting can call asterisk. All devices get registered in opengk. But I
> can't call to any of these from asterisk. I'm defining just
> "Dial,H323/extension_number" in my extensions.conf (the extension number is
> one of the registered in opengk).
> 
> Can anyone help me posting the lines for a basic H323 configuration for
> asterisk?

Here is a sample oh323.conf file:

[general]
listenAddress=0.0.0.0
listenPort=1720
;connectPort=1720
fastStart=yes
h245Tunnelling=yes
h245inSetup=no
inBandDTMF=yes
silenceSuppression=no
jitterMin=20
jitterMax=200
ipTos=none
outboundMax=10
inboundMax=10
gatekeeper=DISCOVER
;gatekeeper=192.168.1.2
userInputMode=TONE
context=voip-h323
;-------------------------------
[register]
gwprefix=6
context=external
gwprefix=069
;-------------------------------
[codecs]
codec=G711U
frames=20


This config file will setup the OH323 channel driver
to use a gatekeeper which will try to discover.
It sets the format of H.323 channels to G.711 ulaw.
The channel driver will also register 2 gateway
prefixes to the gatekeeper: 6 and 069
Incoming calls with called number which start with 6
are routed in context "voip-h323" in extensions file
extensions.conf. Incoming calls with called
number which starts with 069 are routed in context
"external".

Here is a portion of the extensions.conf file:

[phone]
ignorepat => 9
ignorepat => 0
include => parkedcalls
exten => _9XXX,1,StripMSD,1
exten => _200,2,SetCallerID,666
exten => _200,3,Dial,OH323/200 at 192.168.1.122|20|t
;exten => _200,3,Dial,OH323/200|20|t

[voip-h323]
include => parkedcalls
exten => s,1,Goto,i|1
exten => t,1,Playback,demo-thanks
exten => t,2,Hangup
exten => i,1,Playback,pbx-invalid
exten => 666,1,Answer
exten => 666,2,SetMusicOnHold,default
exten => 666,3,Dial,Phone/phone0|30|tH
exten => 660,1,Answer
exten => 660,2,Echo
exten => 660,3,Hangup

[external]
exten => s,1,Goto,i|1
exten => t,1,Playback,demo-thanks
exten => t,2,Hangup
exten => i,1,Playback,pbx-invalid
; Mobile phones
exten => _069XXXXXXXX,1,StripMSD,1
exten => _69XXXXXXXX,2,Answer
exten => _69XXXXXXXX,3,Dial,OH323/BYEXTENSION at 192.168.1.3

Incoming calls to extension 666 ring the Phone/phone0.
Incoming calls to extension 660 initiate the echo test.
Incoming calls to numbers starting with 069 are routed
to a H.323 gateway (192.168.1.3).
Also, the user is able to make outgoing H.323 calls
from a Phone dialing 4-digit numbers starting with a 9.
If the registered number of a Netmeeting is 111, you can
reach it by dialing 9111.

Hope that helps.

> 
> Also, if anyone has a basic SIP configuration for a Cisco router with FXS
> interfaces, it'll be appreciated.
> 
> Regards,
> 
> Carlos Crembil
> Servicios Profesionales
> http://openware.biz
> eMail: ccrembil at openware.biz
> 
> 
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Michael.