Berry The local Telco's have control over the local phone numbers but they make share/collocation/LNP agreements with other carriers and VOIP interconnect carriers so numbers get swapped/leased and rented between different vendors. As a VOIP interconnected carrier this allows us access to 90% of US number markets. If there is a market that we need that one of our partners does not have we try to partner with a player in that region or someone who has. If that does not work we can then collocate equipment with that local carrier to get access. This then extends our network reach to that region. The goal is to achieve the highest quality lowest cost routes to regions our customers are willing to pay for.
---------------------------------------- From: "Nick Khamis" <[email protected]> Sent: Friday, November 04, 2011 10:40 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Subject: Re: [asterisk-users] DID from Direct from Telco Hello Eric, That is also a good idea. I am new to the VoIP world an do not know who the major players are however, will catch on really quick as my background is enhanced neuro networks. I understand all the theory behind compressions, codecs etc... Just trying to apply it in the real world. That being said, I was under the impression that only the local Telcos have control over the phone numbers.I take it that this is not correct? Cheers, Berry. On Fri, Nov 4, 2011 at 10:35 AM, Eric Wieling <[email protected]> wrote: > Why not go direct to Verizon Business (they provide nationwide wholesale SIP > services) or Level3 for your SIP interconnect? Leave the local telco out of > it. > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Nick Khamis > Sent: Friday, November 04, 2011 10:33 AM > To: [email protected]; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] DID from Direct from Telco > > Hello Bryant, > > I just realized how much information Nick has left out. Basically we would > like to function as a DID vendor. > Yes, everything on our end will be converted into SIP using G711 codec . We > have an OC48 coming into our network, and a contact with the local telco here > willing to supply us with a block of phone numbers. The target would be: > > Telco Block of Numbers -> Our Mediant Gateway (E1/T1) -> Our SIP Proxy > -> Customer -> SIP Trunk -> Terminated Call > > As you know the customer could be: > * Another SIP Proxy > * A SIP PBX > > Are E1/T1 mediants capable of handling OC connections? Could you gents > recommend an entry level gateway that could scale? > > Kind Regards, > > Berry. > > -- > _____________________________________________________________________ > -- 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 > > -- > _____________________________________________________________________ > -- 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 > -- _____________________________________________________________________ -- 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
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