Stefan Viljoen
2014-Dec-30 06:22 UTC
[asterisk-users] asterisk-users Digest, Vol 125, Issue 33
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
Lukasz Sokol
2014-Dec-30 09:11 UTC
[asterisk-users] asterisk-users Digest, Vol 125, Issue 33
On 30/12/14 06:22, Stefan Viljoen wrote: ...> > I think you're overcomplicating your problem. (if I understand you > correctly!)This is probably right ;) in both parts> > 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. >Well I did the other way round - I created an app in FreePascal under Windows, entered into onIncomingCall option of MicroSip [btw wrongly described as CSipSimple before] that allows user to copy the number to clipboard and then paste it to the search function of the CRM (web-based - eGropware). Exactly /because/ my phones have no LCD's ... :)> 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. >Yeah, Going To check this 'originate' thingy definitively (Ryan Wagoner also suggested this)> 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?Definitively, going to check this out. Time allowing, probably somewhere in January ;)> > 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? >Alternatively if the MicroSIP softphone could do that 'originate' :) (Advantage: one app less to run in the background)> Thereby avoiding the extremely complicated SIP setup / manipulation you want > to do... > > Just a thought.Thanks, appreciated :)> > Regards > > Stefan > >Kind Regards Lukasz