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viljoens at verishare.... Guest
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Posted: Tue Dec 30, 2014 1:24 am Post subject: [asterisk-users] asterisk-users Digest, Vol 125, Issue 33 |
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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),
.
.
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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
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el.es.cr at gmail.com Guest
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Posted: Tue Dec 30, 2014 4:11 am Post subject: [asterisk-users] asterisk-users Digest, Vol 125, Issue 33 |
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On 30/12/14 06:22, Stefan Viljoen wrote:
...
Quote: |
I think you're overcomplicating your problem. (if I understand you
correctly!)
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This is probably right in both parts
Quote: |
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.
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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 ...
Quote: | 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.
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Yeah, Going To check this 'originate' thingy definitively (Ryan Wagoner also suggested this)
Quote: | 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
Quote: |
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?
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Alternatively if the MicroSIP softphone could do that 'originate'
(Advantage: one app less to run in the background)
Quote: | Thereby avoiding the extremely complicated SIP setup / manipulation you want
to do...
Just a thought.
|
Thanks, appreciated
Kind Regards
Lukasz
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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To UNSUBSCRIBE or update options visit:
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