On 7/24/06, Aaron Anderson <[EMAIL PROTECTED]> wrote:
I have been messing with both all day. I think what might be tripping me up is the extensions.conf.
i do think so too :)
I was able to receive an incoming connection from the client, but all the system returned was a busy signal. This call was to a known good number (my phone) so I'm not sure what's wrong.
if you got the incoming connection but then got a busy tone, it is because the "peer info" in your asterisk conf is not good ... try the "standard" demo setup in asterisk ... there you can call extension 600 (i think) and you get an echo test ... later, extend this to call whoever you want ... but you'll need to enter the "location" info somehow (i never did it ... ). I used sjphone for testing ... try it ... you can add an extension to dial directly to the IP address of the sjphone. So, you dial the MMM extension on your phone, get to asterisk and then asterisk dials the IP address of sjphone ... nice and easy for testing.
Will gnugk do a "translation" from h.323 to sip so I don't have to make any major modifications? Is there an example gnugk.ini and h323.conf file I can look at to get this all running?
gnugk has no clue of sip. gnugk is a gatekeeper for h323 ... you can do stuff with the dialled numbers ... forwarding the call to various [asterisk] h323 gateways and the like ... but the conversion of h323 to sip is done in asterisk. Cesc
Thank you in advance Aaron [EMAIL PROTECTED] wrote: Hello, Try both chan_oh323 and gnugk . Harry --- Aaron Anderson <[EMAIL PROTECTED]> a écrit : I have been scouring the net the last couple of days looking for some kind of tutorial or walkthrough on setting up a h.323 channel in asterisk. What I need to do is basically this: I have a client who wants to be able to connect to me via h.323 and make a local phone call (local to me, he is in a different country). The call is an automated process and no callee interaction is required. My client simply wants to be able to call a user and give them a verification number and then hang up. He's using some in-house software so unfortunately, h.323 is his only option. Can someone point me to a doc or perhaps give me a simple breakdown of what I need to add to asterisk in order to be able to do this? I am on a tight deadline and my searches have not revealed the information I am looking for. I have built chan_h323 and it is loaded but I'm not sure how to set it up beyond that. Any help would be much appreciated. Thank you Aaron _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___________________________________________________________________________ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ----------------------------------------------------------------- This message was scanned by Sunny-Net SpamVirus Detector Gateway If this is not spam but was Maked with SPAM in the subject line please forward this e-mail to [EMAIL PROTECTED] if this is spam but was not marked spam please forward the e-mail to [EMAIL PROTECTED] Thankyou ------------------------------------------------------------ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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