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    Voip Providers for Asterisk

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    • C
      Cheetohz
      last edited by

      Anyone have recommendations for the best VOIP provider for use with Asterisk? I need High Quality and Reliability, as any dropped calls, downtime, or audio anomalies could cost me my job. only one-two lines will be used. primarily for inbound calls, with the occasional outbound call maybe once or twice per month.

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      • E
        Elludium_Q-36
        last edited by

        SOHO, huh?


        Have you considered Google Voice?

        People HAVE integrated it with Asterisk:
        https://www.google.com/search?&hl=en&lr=lang_en&q=%22google+voice+and+asterisk%22&btnG=Google+Search&gbv=1

        The nice thing about GV is, if your Asterisk server, etcetera, goes down, it can be run from a web browser.  Keep in mind, for that to work, in Linux, for example, a plugin needs to be installed:
        https://www.google.com/tools/dlpage/hangoutplugin/download.html

        You can get a "personal" account:
        https://accounts.google.com/SignUp?service=grandcentral&continue=https%3A%2F%2Fwww.google.com%2Fvoice%2F <mpl=open< a="">For a business, you might consider GV, as a part of Google Apps:</mpl=open<> https://www.google.com/enterprise/apps/business/
        That is NOT free, but comes with many useful features…

        Good luck!

        :)

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        • B
          BeerCan
          last edited by

          I use vitelity, voip.ms and teliax and prefer them in that order.  I cannot afford any dropped calls so I have set them up to failover to each other.  I have about 6 DID's and have had minimal problems over the years (about 4 years with these providers).  I use the piaf distro for asterisk but like the freepbx distro as well.

          If you meant a service that hosts asterisk I have heard good things about rentpbx

          IMO google voice is not the best solution as they change things with their service settings, sometimes without warning.

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          • C
            Cheetohz
            last edited by

            Yeah, i was planning to just install the asterisk package in pfsesne and roll with it. looking for a SIP provider, or some way to set up my own usign an analog line

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            • B
              BeerCan
              last edited by

              You will need an fxo card/device if you want to use an analog line.  I am not sure if there are any add in cards with freebsd drivers, when I still had analog lines I used cards from rhino. (http://rhinoequipment.com/analog.aspx)
              If your needs are simple the obi110 might be the ticket. http://www.obihai.com/docs/OBi110DS.pdf  IIRC it is standalone so no drivers needed.

              On a forum I frequent a lot of users have been praising flowroute as a provider.  I never used them but they might be worth investigating.

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              • E
                Elludium_Q-36
                last edited by

                BeerCan is right about GV changing things, unexpectedly.  For example, I use their "Web Call"/Click2Call feature, but don't like that it runs on flash.  A good portion of mobile devices don't support flash, including iPhones.  HTML5 is said to be the fix…  There is a way to make the Web Call feature work, just using HTML4, but, AT ANY TIME, Google can come along and change the coding.

                However, for SOHO environments, it has some aspects that make sense.  It's off-site, they do the work, they store your voicemail and SMS/texts (Yes, texting with what seems like a landline.),  it works with analog / PSTN and has a soft-phone.

                Now, an analog line might seem a bit archaic, but it's not a bad idea as a failover device, in a SOHO environment.  ALL your desktop computers and servers could go down, the power could go out, but an old school RJ-11, single line, corded phone (NOT wireless) would still work.

                There have been issues with FXO/FXS PCI hardware cards where expensive hardware echo cancellation has been MANDATORY.  The software method EATS CPU cycles.

                One failover/failback method, that allows tying an analog line into asterisk, is using something like a Sipura/Linksys spa3000 pstn interface | Linksys SPA3102  .  If the power goes out, for example, the failover is via the analog handset(s).

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                • RicoR
                  Rico LAYER 8 Rebel Alliance
                  last edited by

                  You burrow out some 4 year old thread to do advertising? 🤥

                  -Rico

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                  • stephenw10S
                    stephenw10 Netgate Administrator
                    last edited by

                    Spammers gonna spam! 🙄

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