Compile without SRTP support Hint: ./configure --without-srtp
~ Andrew "lathama" Latham [email protected] * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Sat, Sep 25, 2010 at 1:37 AM, Barry Miller <[email protected]> wrote: > On Fri, Sep 24, 2010 at 10:25:01PM -0700, Ira wrote: >> At 01:14 PM 9/23/2010, you wrote: >> >The Asterisk Development Team has announced the second release candidate of >> >Asterisk 1.8.0. This release candidate is available for immediate download >> >at >> >http://downloads.asterisk.org/pub/telephony/asterisk/ >> >> I downloaded this, ran "./configure" followed by "make menuselect" >> and I don't seem to have SIP as an available protocol. Is there >> something I can do to make it available? It works fine on the most >> recent 1.6 version and it's worked on most of the prior 1.8 versions. > > You probably need to install libssl-dev then rerun ./configure. At > least I did (Debian Lenny). Seems chan_sip needs res_crypto which > needs libssl. > > -- > Barry > > -- > _____________________________________________________________________ > -- 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
