Hi Ryan! Interesting what experience you have made in this issue. We have setup the alternative channel for H.323 (the * built in chan_h323), and we are now in a testing phase.
I was wondering (in case no transcoding is needed), how your setup treats the RTP streams. Do the RTP streams go end-to-end or always via Asterisk? Another question I'd be interested in: Have you also gained some experience with bridging _video_ calls between H.323 and SIP? cheers, Bernie PS: I'd be glad, if I also could get the relevant config files from you. On Fri, 13 Aug 2004, Ryan Wilkins wrote: > Yes, it can.. I'm doing it at my home. My current setup is > Asterisk-1.0-RC2 using the oh323 driver. I have a SIP connection to > Broadvoice talking to Asterisk. I have a e-tel (now Qtelnet) H.323 VoIP > telephone adapter as my end point talking to Asterisk. > > For processing sake, you may want to keep your codec the same all the way > through. Originally I ran G.711u on the SIP connection and G.711a on the > H.323 connection. It worked just fine but the logs always said something > about transcoding between u-law and a-law. I reset the H.323 link to > G.711u and now it says nothing about transcoding. In theory you would > lose a bit of audio quality in the translation process. In reality I > don't really know. > > email me privately if you want a sample config. > > Ryan Wilkins > > > On Fri, 13 Aug 2004, Yiannis Costopoulos, Web2Net Solutions Ltd. wrote: > > > is there a definite answer if asterisk can pass calls between SIP > > and h.323 protocols? _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
