[OpenSIPS-Users] OpenSIPS > announcement > pstn

[Stefan Sayer]

Albert Paijmans wrote:
> 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?
whether you are using SEMS or Asterisk for pre call/early media 
announcement, you would first send the call to the media server of 
your choice, have an announcement played with 183, then the media 
server replies with negative final reply, which you catch in your 
proxy and add as another branch the final destination (pstn/asterisk).

alternatively, you can send the call to SEMS, have the announcement 
played there in early media, and then continue the call in B2BUA mode 
through SEMS (see ann_b2b application, you can modify that a little to 
use 183 instead of 200; or use a simple DSM script and connectCallee 
action).

Regards
Stefan


> 
> Thanks
> 
> Albert
> 
> 
> 
> On Sat, May 15, 2010 at 2:06 AM, Andreas Sikkema <h323 at ramdyne.nl 
> <mailto: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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> 
> 
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-- 
Stefan Sayer
VoIP Services Consulting and Development

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10243 Berlin

tel:+491621366449
sip:sayer at iptel.org
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