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 _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
