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[davide.anzaldi at nete...]

I have the very same situation in one of my networks.
To solve this you can dial out from the softphone and to move call to the
phone you can simply transfer call to the same user (just if you were
transferring call to yourself and the other device will ring.
While, as you notice, you cannot dial a device, you can surely call your
user to tranfer from a device to another.

Please note that call waiting has to be enable on user settings otherwise it
won't work.

This should work both on device/user with chan_sip and pjsip with multiple
devices on same user.

Davide


-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Per conto di Lukasz Sokol
Inviato: lunedì 29 dicembre 2014 13:26
A: asterisk-users@lists.digium.com
Oggetto: [asterisk-users] chan_sip and 2 devices under same extension -
transferring call endpoint(s)

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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