Just as a clarification, FusionPBX is freeswitch not asterisk. On Wed, Sep 23, 2020 at 2:37 PM Daniel White <dwh...@atheral.com> wrote:
> Having a direct connection for a SIP trunk is easy. Just put an Adtran > 904e or something else into the mix. For LNP you don't need to have the > switch on-net... just a device in the exchange for portability and a T-1 or > something else. > > To answer your question... CallerID manipulation is easy right now. In > 9ish months once STIR/SHAKEN is a thing... you may run into issues getting > calls signed with a TN for CallerID that isn't on that providers service. > > So yes... to answer your question it is easy to do on an Asterisk based > system (FreePBX or FusionPBX for instance) but someone like Lewis and I > could save you a lot of trouble for certain. That and it is easy today... > but is going to be a lot more complicated next year. > > [image: photograph] > Daniel White > Co-Founder > phone: +1 (702) 470-2770 > direct: +1 (702) 470-2766 > > ch...@wbmfg.com > September 23, 2020 at 12:52 > The provider charges more for a “trunk”. > And Lewis, I already have this service set up through them for another > system > We have some unique LNP porting issues and really want the voip provider > to have a direct connection to us. > > *From:* Brian Webster > *Sent:* Wednesday, September 23, 2020 12:35 PM > *To:* 'AnimalFarm Microwave Users Group' > *Subject:* Re: [AFMUG] OT sip question > > > It’s been a while since I have done an asterisk config but you still > configure a SIP account much the same as you would for the ATA and it > becomes a trunk as I recall and then you do whatever you need to with it as > a single line trunk. Way back when I was learning it and playing I had > multiple providers for single lines and even an IAX line too. Then I could > use them as part of the dial plan and other features as to where it would > ring and dial out. > > > > Thank you, > > Brian Webster > > www.wirelessmapping.com > > > > *From:* AF [mailto:af-boun...@af.afmug.com <af-boun...@af.afmug.com>] *On > Behalf Of *ch...@wbmfg.com > *Sent:* Wednesday, September 23, 2020 2:01 PM > *To:* af@af.afmug.com > *Subject:* [AFMUG] OT sip quiestion > > > > If I get a voip service from some random provider and I have a sip device > on my end, I am golden. Say an ATA connected to a pots phone. > > Now, if I want that number to appear inside an asterisk PBX but I don’t > want to buy a trunk, is there an easy way to do that? > > ------------------------------ > -- > AF mailing list > AF@af.afmug.com > http://af.afmug.com/mailman/listinfo/af_af.afmug.com > Brian Webster <i...@wirelessmapping.com> > September 23, 2020 at 12:35 > > It’s been a while since I have done an asterisk config but you still > configure a SIP account much the same as you would for the ATA and it > becomes a trunk as I recall and then you do whatever you need to with it as > a single line trunk. Way back when I was learning it and playing I had > multiple providers for single lines and even an IAX line too. Then I could > use them as part of the dial plan and other features as to where it would > ring and dial out. > > > > Thank you, > > Brian Webster > > www.wirelessmapping.com > > > > *From:* AF [mailto:af-boun...@af.afmug.com <af-boun...@af.afmug.com>] *On > Behalf Of *ch...@wbmfg.com > *Sent:* Wednesday, September 23, 2020 2:01 PM > *To:* af@af.afmug.com > *Subject:* [AFMUG] OT sip quiestion > > > > If I get a voip service from some random provider and I have a sip device > on my end, I am golden. Say an ATA connected to a pots phone. > > Now, if I want that number to appear inside an asterisk PBX but I don’t > want to buy a trunk, is there an easy way to do that? > > > > -- > AF mailing list > AF@af.afmug.com > http://af.afmug.com/mailman/listinfo/af_af.afmug.com > -- Lewis Bergman 325-439-0533 Cell
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