[Asterisk-Users] Dialling a DUNDi Route

Douglas Garstang dgarstang at oneeighty.com
Thu May 11 09:33:56 MST 2006


> 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.






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