This is only true for PRI, T-1 and other PSTN services. The wholesalers take care of all everything to do with the PSTN side and number ports, etc. Also check out Gafachi and Vitelity for service on a smaller scale. Level3 and Verizon have some hefty mins/month commitments.
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Friday, November 04, 2011 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion 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 <ewiel...@nyigc.com> 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: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick > Khamis > Sent: Friday, November 04, 2011 10:33 AM > To: brya...@zktech.com; 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 -- _____________________________________________________________________ -- 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
