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I am my own provider: incoming DIDs reside with a wholesale SIP trunking service. Outgoing SIP with the same provider, usually, though I have multiple possible
by walrus01 5y ago
I am my own provider: incoming DIDs reside with a wholesale SIP trunking service. Outgoing SIP with the same provider, usually, though I have multiple possible routes. The asterisk system, I run myself. My outgoing caller ID (and my carrier's links to its upstreams such as bandwidth.com) are SHAKEN/STIR compliant.
- aeontech 5y agoThis sounds fascinating and also completely over my head in terms of possibilities… is there a basic “getting started guide” you would recommend for someone who may not have the bandwidth to go all-in learning telephony from soup to bolts but would still like the benefits? :)
- walrus01 5y agoWouldn't recommend it unless you do network/telecom stuff for a living, it has a bit of an ongoing time cost in maintenance... some of the SIP trunking providers that have more focus on the consumer market such as voip.ms do let you define your own custom IVR and menus, to accomplish pretty much what I described, without having your own SIP-speaking system registered to it persistently. You should be able to define a voip.ms IVR destination as your cellphone.
- jasonjayr 5y agoThinking about this a smidge, you probably can just use voip.ms to (a) order a DID (aka a phone number), (b) setup an IVR for the message, and (c) instruct them to dial an [internal to your account] extension that forwards to your real number, all on the voip.ms platform. You wouldn't have to set anything up, just pay them for per-minute charge. You will have to pay 2x per minute, once for the inbound, and then outbound to your real number, ..... unless you go through the extra steps of setting up SIP phones. FWIW; I've used voip.ms for a long time, and just a very happy customer. The biggest trouble they've had was availability issues a month or so ago when they were the victim of a DDoS :-/. You probably could pull off this trick with any of the pay as you go SIP trunking providers ...
- gruez 5y agoHow do the calls actually get routed to your phone? Via SIP, or through your cell provider (ie. it dials your phone's phone number)? If everything is over SIP, then it makes sense that everything is SHAKEN/STIR compliant.
- walrus01 5y agoat present I do it as the latter. My cellphone's direct number is an external destination on the asterisk system and it initiates a new outbound call to it, and bridges the audio/call from the incoming caller to it. I can also do it direct over SIP if I want. The android linphone SIP softphone client is not bad. My cellphone can register to my asterisk system as the same extension as my SIP desk phone in my home office. Or if I want I can give it a new extension number and make my desk phone extension number, and the linphone softphone extension number part of the same ring group for me as a person. In which case I would send x500 from the IVR main menu to that ring group. I generally don't leave linphone running and connected unless I'm actively using it because it can be a battery/CPU hog (in order to SIP register to my asterisk system I also need to be on my personal openvpn link, my server doesn't accept incoming connections from any public IP on the internet). So a combination of the openvpn client running and linphone simultaneously.
- gruez 5y ago> at present I do it as the latter. My cellphone's direct number is an external destination on the asterisk system and it initiates a new outbound call to it, and bridges the audio/call from the incoming caller to it. But under that scheme, would it still work SHAKEN/STIR? I would think not, because to your cell provider it's effectively indistinguishable from a bad guy spoofing caller ids. Or perhaps there's some forms or whatever you can fill out to get a certificate that allows a third party to spoof calls to you?