Hi Tim, it seems that using trunks is the right way....is this what you meant?
Tim Panton wrote: > Use IAX :-) > > In principle chan_skype could also support it. > > T. > > On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: > >> Hi, >> >> Does anybody knows where I can find some docs about how to make the URL >> parameter inside the Dial command work? I tried to make some tests with >> a sip phone without success: the sip debug shows no URL inside sip >> packets. :( >> Any hint appreciated. :) >> >> Thank you >> >> Giorgio >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > Tim Panton - Web/VoIP consultant and implementor > www.westhawk.co.uk > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _________________________________________________ Giorgio Incantalupo, mailto:[email protected] FG&A srl - http://www.fgasoftware.com - vo...@work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
