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/[EMAIL PROTECTED]|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/[EMAIL PROTECTED]
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: [EMAIL PROTECTED]
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Michael.
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