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:-). 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? Regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh@imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060706/459cf7dc/attachment.htm
Apollon Koutlides
2006-Jul-06 01:57 UTC
[asterisk-users] B2BUA Webbased and Click 2 dial apps
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 : dinesh@imcb.a-star.edu.sg <mailto:dinesh@imcb.a-star.edu.sg> > 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
Marnus van Niekerk
2006-Jul-06 01:58 UTC
[asterisk-users] B2BUA Webbased and Click 2 dial apps
Also have a look at .call files. You web app can just create a .call file and then move it to the right location and asterisk will place the call No manager interface needed. Marnus van Niekerk "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. 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 niceJ. 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? > > > > Regards, > > Dinesh Birlasekaran > Network Engineer, > ComIT, Institute of Molecular and Cell Biology > 61 Biopolis Drive, Singapore 138673 > HP : 92962676 DID : 65869804 Fax : 67791117 > Email : dinesh@imcb.a-star.edu.sg <mailto:dinesh@imcb.a-star.edu.sg> > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060706/8da76ba3/attachment.htm
It would be hard to bill all this calls, if you are using dialout call files instead of Asterisk Manager API no ? How would you colect the call duraction of both call legs? Thks, Marco Mouta On 7/6/06, Marnus van Niekerk <m@mjvn.net> wrote:> > Also have a look at .call files. > > You web app can just create a .call file and then move it to the right > location and asterisk will place the call > No manager interface needed. > > Marnus van Niekerk > "Opportunity is missed by most people because it is > dressed in overalls and looks like work." > > Thomas Alva Edison - Inventor of 1093 patents, > including the light bulb, phonogram and motion pictures. > > > > 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 niceJ. 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? > > > > Regards, > > Dinesh Birlasekaran > Network Engineer, > ComIT, Institute of Molecular and Cell Biology > 61 Biopolis Drive, Singapore 138673 > HP : 92962676 DID : 65869804 Fax : 67791117 > Email : dinesh@imcb.a-star.edu.sg > WWW: 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 > > >-- Com os melhores cumprimentos, Marco Mouta
I am not for the billing part, as its sip based, and its educational calls only. I mean between sip.edu community and my educational institute. So practically any sip uri should be able to be dialed from the website. I dunno I am just asking the ideas for the group. Regards, Dinesh. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Marco Mouta Sent: Friday, July 07, 2006 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] B2BUA Webbased and Click 2 dial apps It would be hard to bill all this calls, if you are using dialout call files instead of Asterisk Manager API no ? How would you colect the call duraction of both call legs? Thks, Marco Mouta On 7/6/06, Marnus van Niekerk <m@mjvn.net> wrote:> > Also have a look at .call files. > > You web app can just create a .call file and then move it to the right > location and asterisk will place the call > No manager interface needed. > > Marnus van Niekerk > "Opportunity is missed by most people because it is > dressed in overalls and looks like work." > > Thomas Alva Edison - Inventor of 1093 patents, > including the light bulb, phonogram and motion pictures. > > > > 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 bridgethe> call? Do I need any special sip api for this? Any ideas will be niceJ.Does> this webpage has to be on asterisk server running on the machine? Or canit> be passed as a string to the server from the webserver? > > > > Regards, > > Dinesh Birlasekaran > Network Engineer, > ComIT, Institute of Molecular and Cell Biology > 61 Biopolis Drive, Singapore 138673 > HP : 92962676 DID : 65869804 Fax : 67791117 > Email : dinesh@imcb.a-star.edu.sg > WWW: 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 > > >-- Com os melhores cumprimentos, Marco Mouta _______________________________________________ --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