Hi All; The asterisk version I am using is 1.8.4.2 and I compiled ooh323 channel (by selecting the add-on). But really does not work in good performance, for example: if a call came from gnugk to asterisk and the ooh323 handled it, the performance is bad .. some calls are drop and if it is ringing, then it rings for small duration and then stop ringing ....
In other words, if the call went from gnugk to the provider directly (all the path h323), it is better than coming for Asterisk via the ooh323 channel and then to be translated for SIP to be sent for provider. I would like to try the h323 channel (and not the ooh323), but I do not know what I have to do to compile? Any advise? Did anyone tried yate to do the translation from h323 to sip? How it is? Regards Bilal -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
