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? 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@openware.biz
Sorry, my dial line is "Dial,OH323/extension_number"... Regards... Carlos Carlos Crembil/Openware/AR@OPENWAR Para: <asterisk-users@lists.digium.com> E cc: Enviado por: Asunto: [Asterisk-Users] SIP Model and H323 asterisk-users-admin@lists. digium.com 17/03/2003 11:36 p.m. Por favor, responda a asterisk-users 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? 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@openware.biz _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
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@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@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@openware.biz > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-usersMichael.
Thank you Michael!. I've applyied the configuration you sent me, but I have some troubles with it, specially in the oh323.conf file. Lines like "[register]", "[codecs]" are not appearing in my original file, and when I use this, asterisk returns me an error and it fails to start. It also happens with lines like "fastStart" (in my original file, this line is "noFastStart=no"). Exist more than one OH323 driver? Regards, Carlos Carlos Crembil Professional Services http://openware.biz eMail: ccrembil@openware.biz Michael Manousos <manousos@inaccessnetworks. Para: asterisk-users@lists.digium.com com> cc: Enviado por: Asunto: Re: [Asterisk-Users] SIP Model and H323 asterisk-users-admin@lists. digium.com 18/03/2003 09:16 a.m. Por favor, responda a asterisk-users 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 numberis> 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". ....> > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-usersMichael. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Carlos Crembil wrote:>Exist more than one OH323 driver? > >Yes . You can find the other one at http://asterisk.nufone.net/ Jeremy McNamara
Ok! Thank you! Carlos Crembil Professional Services http://openware.biz eMail: ccrembil@openware.biz Jeremy McNamara <jj@indie.org> Para: asterisk-users@lists.digium.com Enviado por: cc: asterisk-users-admin@lists. Asunto: Re: [Asterisk-Users] SIP Model and H323 digium.com 20/03/2003 03:55 p.m. Por favor, responda a asterisk-users Carlos Crembil wrote:>Exist more than one OH323 driver? > >Yes . You can find the other one at http://asterisk.nufone.net/ Jeremy McNamara _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users