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[el.es.cr at gmail.com]

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),

I have several extensions that can register 2 separate devices (chan_sip)
( FreePBX calls this Devices & Users mode : Users are extension/internal number,
devices are the 'SIP Accounts' for the internal 'endpoints' )

(this I'm told apparently will not be needed if I switched to chan_pjsip,
since it allows multiple devices to register on the same user/secret,
so the u/d mode would not make sense any more; however this creates another interesting problem, pls read on)

Some endpoints are grouped in pairs so that calling an extension, rings on both devices.

(One 'device' is a real handset, usually dumb: SPA112 or SPA301,
the other is a softphone (CSipSimple or WebRTC or both) used to bring the incoming CID to
users' eye level and to perform some client-side CRM integration )

On Incoming call, as expected, the softphone shows me the CID [as intended]
and I can pick up the handset, then the softphone will stop ringing;
This far, it works as intended and no problems here.

I *think* by the FreePBX convention (?) one can not call the 'device' number/reg directly,
only the 'user' extension [i actually tried dialing to one of the 'device' SIP reg numbers,
'cannot be completed as dialed' was the answer, and same in the -vvvvr output;
the -vvvvr output actually suggests one side RTP is passed, but the other is not,
if I read this correctly (on 'normal' calls, both sides RTP is shown 'passed' in the log).

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 ?

An answer from the FreePBX forum suggested SLA / Shared Line Appearance - but
as I read description of that, it's not really: there is no master/slave in the pair,
both devices are *supposed* to be of 'equal rights' as they are 'manned' by the
same person. IOW my use case is *simpler* than SLA...

The interesting question also is how would one do this with chan_pjsip,
if a user can have multiple devices registered on the same 'SIP Account',
how could the user 'transfer the call endpoint' between his devices
(whether the call is incoming or outgoing) ?

Hope the above makes (some) sense,

Kind Regards



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[rswagoner at gmail.com]

On Mon, Dec 29, 2014 at 7:26 AM, Lukasz Sokol <el.es.cr@gmail.com (el.es.cr@gmail.com)> wrote:

[el.es.cr at gmail.com]

On 30/12/14 02:52, Ryan Wagoner wrote:
[...]

[drdialtone at optonlin...]

Sounds like a job for TAPI.
Google TAPI for Asterisk or Asterisk TSP
I've been playing with SIPTAPI and it works pretty well.  It's very simple to install and set up.
Hope this Helps
PC...

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ryan Wagoner
Sent: Monday, December 29, 2014 9:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] chan_sip and 2 devices under same extension - transferring call endpoint(s)


On Mon, Dec 29, 2014 at 7:26 AM, Lukasz Sokol <el.es.cr@gmail.com (el.es.cr@gmail.com)> wrote:

[el.es.cr at gmail.com]

On 30/12/14 12:21, Pat Collins wrote: