Re: [asterisk-users] Click2call from an OpenOffice document
Make a html link this way: http://widget.doddlephone.com/embed/webphone.jsp?sipserver=SERVER&username=USER&password=PASSWORD&callto=PHONETOCALL&auto=yes "> Tel: +1 234 567 890 /b Sergio On Fri, Jul 9, 2010 at 5:29 AM, Olivier wrote: > Hi, > > What would you suggest to get click2call from an OpenOffice document ? > For instance, in OOo Writer, there is a block : > > M. John Doe > Tel: +1 234 567 890 > email: j...@example.com > > Looking at this block, the line +1 234 567 890 is underlined. > When clicking on this, a contextual menu pops up allowing you to make a > call. > > Regards > > -- > _ > -- 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
Re: [asterisk-users] Software for PC-PC voice comunication
Hi, If you want an online option to make calls right from webpage, you can use doddle online SIP webphone: http://widget.doddlephone.com/ Sergio On Wed, Oct 28, 2009 at 11:11 AM, Zoa wrote: > > Give zoiper a try, http://www.zoiper.com (I'm working for them) > Works with SIP and IAX, and should be pretty easy to setup. > > Zoa > > > giancarlo lombardo wrote: > > I just installed an Asterisknow server > > can someone suggest a software to be used for a PC - PC voice > > comunication to test in easy way the functionalities of my server. > > > > Thanks in advance for the help > > > > > > ___ > > -- 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 > > > ___ > -- 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 > ___ -- 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
Re: [asterisk-users] Asterisk,click2talk, webphone
Hello We have launched another Gadget Option (Call Extension Button) to use with Asterisk: For instance: extensions.conf: [callButtonRoute] exten => 444,1,Dial(SIP/18874...@10.0.0.1,90,t) exten => 444,2,Hangup sip.conf: [callButton] type=peer username=callButton secret=1qa2ws3ed context=callButtonRoute host=dynamic nat=yes disallow=all allow=alaw allow=gsm Doddle Call Extension Button Gadget: http://www.doddling.com/doddle/embed/doddleEmbedded.jsp?sipserver=10.0.0.1&username=callButton&password= 1qa2ws3ed&callto=444&label1=My%20Asterisk&label2=Hangup%20Now&auto=yes&mode=button" width="243" height="160" frameborder="0" scrolling="no" > You can check all URL parameters options at: http://www.doddplephone.com Regards Sergio On Fri, Oct 2, 2009 at 7:38 PM, Doddle WebPhone wrote: > Hi,This can be useful for Asterisk / TI integrators: > > How to create a Free click2call application using Asterisk: > > We can build click2talk / webphone application empowering webpages with > VoIP Telephony using online DoddlePhone and Asterisk > > Invoke doddle webphone (http://www.doddlephone.com) as follows: > > sipserver=Asterisk_SERVER&username=USER&password=PASSWORD&callto=PHONE_NUMBER_TO_CALL > > > Just create sip account on Asterisk, define its route and trigger > click2call as above > Notice that we can set a fixed route / context to the click2talk sip peer. > > > check out http://www.doddlephone.com for details > > Regards > Sergio > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk,click2talk, webphone
Hi,This can be useful for Asterisk / TI integrators: How to create a Free click2call application using Asterisk: We can build click2talk / webphone application empowering webpages with VoIP Telephony using online DoddlePhone and Asterisk Invoke doddle webphone (http://www.doddlephone.com) as follows: sipserver=Asterisk_SERVER&username=USER&password=PASSWORD&callto=PHONE_NUMBER_TO_CALL Just create sip account on Asterisk, define its route and trigger click2call as above Notice that we can set a fixed route / context to the click2talk sip peer. check out http://www.doddlephone.com for details Regards Sergio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
