Found this and it works: https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport
On Fri, Sep 12, 2014 at 1:31 AM, Rusty Newton <[email protected]> wrote: > On Wed, Sep 10, 2014 at 2:14 PM, Rizwan H Qureshi > <[email protected]> wrote: > > Hi Everyone, > > How can I create a TLS based sip trunk between two asterisk servers. I > have > > been trying to do it but i dont know how to defined the client > certificate > > on the asterisk server. Has anyone tried this? > > There is a tutorial for secure calling with TLS and SRTP here: > https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial > > -- > Rusty Newton > Digium, Inc. | Community Support Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct: +1 256 428 6200 > > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- 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 > -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi
-- _____________________________________________________________________ -- 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
