On May 15, 2010, at 3:35 AM, Albert Paijmans wrote: > Thanks for the reply. The reason we do not want to use Asterisk, but SEMS, is > because SEMS offers the possibility to play a different announcement (could > be from database) to every extension. This ofcourse makes it more attractive > to our sponsors. We want to do both sponsor messages for outgoing calls and > we will have some discreet advertisement on our website. We think we can > offer free phonecalls to most international destinations thanks to Open > Source and we are all volunteers :) > > So forwarding calls to Asterisk and using Asterisk as a media server for > voicemail or busy tones I understand that part. But how could I send outgoing > (pstn) calls to SEMS first and then to Asterisk? Is there something like a > service route for this?
The dialplan in Asterisk is much much much more flexible than a lot of people seem to realize. It's in some ways quite a powerful programming language, although it does have weaknesses (some larger than others). And since you already have an Asterisk in the callpath it seems to me to be superfluous to add another element, that will just make things a lot less reliable. You can do conditional branching and database queries from the dialplan, that's all the power required to create a variable experience for each call. It just takes a little lateral thinking and some tinkering. If you want to you could use an AGI script, but I always feel like that being a cop-out, it's more fun to do it from the dialplan. Now, let's get back to OpenSIPS ;-) -- Andreas Who has at one time created a voicemail system in the asterisk dialplan without using the existing voicemail fucntionality _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
