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| uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed. |
| Tags: call , forward , phone , sip |
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#1
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| Don't have one I can try this on, so could someone supply the answer please? If you have a SIP-based service from a remote Asterisk server, and set Call Forward, Always to a mobile (say). Where does the 'forward' take place? Are you sending instructions back to the server, which then does the divert at that end until told otherwise, or does the call actually come into the phone, and back out again? What I would like to achieve, is to have an IP-based phone in the Office, divert to mobile whilst travelling. Take the phone to a remote office, and be able to take the divert/forward off from there. TIA, Philip Partridge |
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#2
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| One idea might be to create a call queue in Asterisk just for Phil's use, where calls to Phil's DDI go. That way, you can log in to the queue from wherever location you have a SIP phone and take calls on your DDI. And set it up so that when you're not logged in to your queue, calls get diverted to your mobile. That way the Asterisk server will know instantly that you're not available as soon as you log out of your personal queue, and won't try to make a connection to a phone that isn't there. As for how the divert actually works; Something tells me mobile telcos aren't quite progressive enough yet to offer you an IAX interface to your mobile, so you'd have to route the divert to your mobile over the PSTN, the same way you'd place calls to any other PSTN number. It would make most sense to set the divert up on the Asterisk server rather than on a SIP phone. alexd |
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#3
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| One idea might be to create a call queue in Asterisk just for Phil's use, where calls to Phil's DDI go. That way, you can log in to the queue from wherever location you have a SIP phone and take calls on your DDI. And set it up so that when you're not logged in to your queue, calls get diverted to your mobile. That way the Asterisk server will know instantly that you're not available as soon as you log out of your personal queue, and won't try to make a connection to a phone that isn't there. As for how the divert actually works; Something tells me mobile telcos aren't quite progressive enough yet to offer you an IAX interface to your mobile, so you'd have to route the divert to your mobile over the PSTN, the same way you'd place calls to any other PSTN number. It would make most sense to set the divert up on the Asterisk server rather than on a SIP phone. alexd |
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