On 06/08/12 13:48, Daniel-Constantin Mierla wrote: > * http://asipto.com/u/68 > > The tutorial focuses on how to use Asterisk's database structure to > perform authentication in Kamailio SIP server, along with user location, > nat traversal, instant messaging, presence, a.s.o., offloading > processing from Asterisk. Asterisk will still handle all the calls, > enabling rich telephony such as MoH, transcoding, ring back, IVR, etc.
This is a good tutorial, but can you clarify the scope of what Kamailio will do in this configuration? - just scalability and protocol conversion (e.g. UDP with Asterisk, TLS with phones)? - does it mean Kamailio is also intended to add other services, e.g. presence and IM functionality? - any comments on using the Jabber gateway module? - is it intended for fully federated SIP, e.g. someone sets this up for example.org, and somebody else in example.com can make a call to [email protected], routed over the public Internet, using DNS SRV and mutual TLS? If it is intended that someone can turn on the mutual TLS mode and use it to federate their Asterisk server, then I'd like to link to it from http://www.opentelecoms.org/use-a-sip-proxy-instead-of-asterisk -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
