chan_h323 looks pretty good for me. It has some bugs though, We are working on issue http://bugs.digium.com/view.php?id=9299 we have some progress there http://voipsolutions.ru/asterisk_segfault_in_chan_h323_under_heavy_load_20080226 I suppose it would be fixed very soon.
Dovid Bender wrote: >> Hello, >> >> Has anyone used Asterisk for IAX - SIP - H323 Protocol Translation all in >> the same box and in production? >> >> If yes, what have you used for H323 part? I'm not concerned about RTP >> Passing through the Asterisk Box (except maybe for IAX), and it is not >> used >> as an User Agent. >> >> I want to know has it worked in a SoftSwitch Situation for Signal Proxy >> and >> Protocol Conversion? and if yes, how?! >> >> Thanks for your enlightenment, >> >> Seysan >> -------------- next part -------------- >> > Seysan, > I would not reccomend using H323 with asterisk (atleast not ooh323). I found > on tests that it would core dump a few times a day. I had this issue in both > 1.4.X and 1.2.X. Have a ook at some Patton boxes. > > Dovid > > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-biz > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
