[OpenSIPS-Users] OpenSIPS > announcement > pstn

[Albert Paijmans]

Hi,

We want to be able to sponsor our outgoing calls to pstn in order to make
these calls free. So when a number starting with 00, 0 or + the call will
first be forwarded to an announcement server (SEMS). Then the call will
continu on to Asterisk.

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 am able to route calls to Asterisk but how do I add an extra server to
play an announcement? Could you please advise where to look?

Thanks,

Albert
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