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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
