(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,