[asterisk-users] chan_sip and 2 devices under same extension - transferring call endpoint(s)

Lukasz Sokol el.es.cr at gmail.com
Mon Dec 29 06:26:12 CST 2014

(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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