On Thu, Jan 31, 2019, 9:24 AM Jean-Denis Girard <jd.gir...@sysnux.pf wrote:
> Hi list, > > Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a > system that uses exclusively tel: uri on inbound and outbound calls. I > could not find documentation or sample config about tel:uri. Is this > doable? If not possible with PJSIP, is chan_sip a better option? Any > pointer would be greatly appreciated. > Right now, chan_pjsip does not properly handle tel: URIs. If you need them you might need to use chan_sip. Matthew Fredrickson > > Thanks, > -- > Jean-Denis Girard > > SysNux Systèmes Linux en Polynésie française > https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users