[OpenSIPS-Users] OpenSIPS Server configuration ( SIP Server ) based on VOIP

[Bogdan-Andrei Iancu]

You can use a PSTN termination service (for both inbound and outbound 
PSTN calls) from a PSTN wholesale provider - you will simply send to 
them the traffic that needs to be terminated on PSTN.  For inbound 
traffic, the PSTN provider will send you (as SIP) the calls targeting 
your DIDs (you need to register the DIDs to the provider).

So, you can either use such a service, either set up your own GW 
(ASterisk, Yate, etc)....But this requires you to buy/rent some lines 
from a telco.

Regards,
Bogdan

gigastar wrote:
> Thanks for your response. 
>
>        Gateway in the sense, is their a need to configure it from our end or
> can we get GW server from the ISP's end.
>   
>        If GW server  has to be configured from our end can we install the
> asterisk in same server ie in OpenSIPS  
>
>        installed m/c and integrate both the s/w's basing the link
> http://www.opensips.org/Resources/DocsTutAsterisk.
>
>        Will this be enough for making outgoing calls. 
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro