> Douglas Garstang wrote: > > We're doing all of our call routing from a database accessed from > > AGI. When we trunk calls from one asterisk system over to > another via > > IAX to terminate the call, the dialling parameters are defined by > > what's in the dial command on the second system, not the first. This > > is a big problem. :( > > Errrrh, ok, I have a very faint idea of what you are saying. But what > are you trying to achieve ?
What am I trying to achieve? Uhm... a carrier grade, highly redundant (ie multiple servers), VOIP solution with advanced business(not residential) features such as findme/followme, incoming and outgoing blacklisting/whitelisting(user/org/company level), user/prefix defined pic codes and rate centers, intra company 4 digit extension dialling, feature codes, user defined internal, external, override caller id and on and on - all provisionable and maintainable via a web interface (don't forgot the multiple servers!)!.... does that answer your question? When you IAX trunk a call from Asterisk A to Asterisk B, you can't pass the ring time and ring options of the original SIP call between servers. Doug. _______________________________________________ --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
