[OpenSIPS-Users] OpenSIPS > announcement > pstn

[Andreas Sikkema]

On May 14, 2010, at 11:13 PM, Albert Paijmans wrote:

> Is it possible to add an extra announcement server in the call path?
> So OpenSIPS acts as registrar/proxy, Asterisk does pstn, voicemail etc. But on certain destinations the call is relayed through an announcement server before continuing to Asterisk.

I'd just use the existing Asterisk for it (providing it has a reliable timing source) and have it play a wav file during "ringing phase" and after the WAV file ends do the rest of the dialplan and have the outgoing call answer the incoming call.

This sudden influx of "let's do add before the call" business plans of late really takes me back to my first VoIP operator job, they just stopped doing that (in the Netherlands and Germany) because there was no money around 2002 after the whole 9/11 thing when there was an economic crisis and advertisers stopped advertising  ;-)

I must be getting old....

-- 
Andreas