Dinesh wrote: > > > Hello, > > > > I have a requirement of bridging 2 sip connections via asterisk, which > has to be web based. > > > > A person has to go to a webpage and enter his from sip uri(say sip1) and > enter another sip uri(say sip2). Upon pressing the connect button, the > webpage needs to send say a dial sip1 uri and dial dip uri 2 and bridge > the call? Do I need any special sip api for this? Any ideas will be nice > J . Does this webpage has to be on asterisk server running on the > machine? Or can it be passed as a string to the server from the webserver? >
The easiest way to implement this is by placing a call-spool file in the proper directory - usually /var/spool/asterisk/outgoing, depends on your asterisk setup. More details here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out You can make this work locally (your cgi/jsp/whatever runs on the same box as the asterisk daemon) - or via nfs, ftp or a million other 'dirty-hack' paths; but there's also another, much cleaner approach using the Asterisk Manager Interface 'Originate' action: http://www.voip-info.org/wiki/index.php?page=Asterisk%20Manager%20API%20Action%20Originate > > > Regards, > > Dinesh Birlasekaran > Network Engineer, > ComIT, Institute of Molecular and Cell Biology > 61 Biopolis Drive, Singapore 138673 > HP : 92962676 DID : 65869804 Fax : 67791117 > Email : [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > WWW: www.imcb.a-star.edu.sg <http://www.imcb.a-star.edu.sg> > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
