[OpenSIPS-Users] OpenSIPS > announcement > pstn

[Albert Paijmans]

Hi Andreas,

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?

Thanks

Albert



On Sat, May 15, 2010 at 2:06 AM, Andreas Sikkema <h323 at ramdyne.nl> wrote:

> 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
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