Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...)
i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have no LCD's to show the dialled number, I want to give the workforce the ability to dial OUT using the softphone, (as in, copy/paste the number from the CRM software into softphone then *immediately* transfer the originated call 'endpoint' to the handset of the same 'user' extension, somehow, the question is, HOW ? --- I think you're overcomplicating your problem. (if I understand you correctly!) Your scenario is almost exactly ours, except we use ATCOM-820P's (with LCD displays) and no softphones. So incoming CID is displayed on the phone's physical LCD displays. What we did is write our own C# dialler app - all this effectively does (through a third-party server app we designed) is connect over the AMI to the Asterisk instance and then use the "originate" function to originate a call to the user's phone. Behind this is a database where we store which logged in user in the dialler app is which extension - e. g. by updating the DB we can "send" a call originated by one user "anywhere" among the group of SIP phones connected to the Asterisk. E. g. I think you can do this too? Instead of them copying the number into the softphone (causing all your SIP pain / confusion to get the "real" phone to then ring with an outgoing call queued to that number) have a second app running (it can be TINY - both in amount of code and on-screen presence) - that does an AMI originate with the Asterisk and sends the desktop originated call to the relevant hardphone? Thereby avoiding the extremely complicated SIP setup / manipulation you want to do... Just a thought. Regards Stefan -- _____________________________________________________________________ -- 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
