They don't require authentication of invites which I do need
Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini <[email protected]> wrote: > 2012/3/22 Zohair Raza <[email protected]> > >> Hi, >> >> How to allow registered sip users to call without re-authentication >> >> insecure =yes/very are deprecated in 1.8 >> >> I want to avoid fromuser= in peer configuration. When I add this in peer >> asterisk, my asterisk accepts call otherwise it says username mismatch. >> >> Please help >> >> >> Regards, >> Zohair Raza >> >> > There are other options, like "invite" and "port" to be used when you > trust the IP of the caller. > > Leandro > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
