Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can paint a picture on how the DID suppliers function it would be greatly appreciated. If I were to guess it would be: Telco Lines -> Gateway E1/T1 -> SIP Proxy -> Media Servers? With this scenario, do we now have control over the number of channels? Thanks in Advance, Nick.
On 11/03/2011 07:20 PM, Nick Khamis wrote:> Hello Everyone, > > Unlike going through DIDx, DIDLogic etc.., we have an option of > getting DIDs directly > from local telco Bell Canada. Currently our SIP Trunk provider > assigned a DID to us, > and as you know, they just redirect requests it to our PBX. > However, when dealing directly with a telco, what equipment will we > need? Basically > giving us the same capability as a DID provider. If someone can paint > a picture on how > the DID suppliers function it would be greatly appreciated. > > If I were to guess it would be: > > Telco Lines -> Gateway E1/T1 -> SIP Proxy -> Media Servers? > > With this scenario, do we now have control over the number of channels? > > Thanks in Advance,Simplest (with 3-4 T1s): Telco Lines -> Asterisk box with T1 card (and possibly a codec processor card) -> Customer More complex (with a bunch of circuits) : Telco Lines -> Gateway T1 -> SIP Proxy -> Media Servers -> Customer And if your question of "number of channels" is "Can I control the number of channels a customer can use simultaneously?", then the answer is "With Asterisk, Yes"
Hello James, Thank you so much for your response. We just purchased an AudioCodes MP124 for this job. And setting up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the Telco here in Toronto. As for other Telcos around the world, for example Bell South in the states, is it possible to have them route a block of Florida phone numbers to our FXS port here in Canada, or do we have to have a T1 gateway + SIP Proxy in Florida, routing the calls to our setup in Toronto and vice versa? Thanks in Advance, Nick.
On 11/03/2011 09:16 PM, Nick Khamis wrote:> Hello James, > > Thank you so much for your response. We just purchased an AudioCodes > MP124 for this job. And setting > up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the > Telco here in Toronto. As for other > Telcos around the world, for example Bell South in the states, is it > possible to have them route a block of > Florida phone numbers to our FXS port here in Canada, or do we have to > have a T1 gateway + SIP Proxy in Florida, > routing the calls to our setup in Toronto and vice versa?Routing Florida numbers up to Canada would get you charged LD per minute fees. You can go with a provider like Level 3 or Global Crossing and they can hand you a T1 circuit that has DIDs from many different areas in the US.
Fair enough, In regards to the the diagram discussed earlier: Telco Lines -> Gateway T1 -> SIP Proxy -> Media Servers -> Customer I understand that a T1 Gateway that has 480 channels, can handle up to 240 calls. That is more than enough for the "Gateway T1 -> SIP Proxy" part of the diagram. I just want to make terribly sure I understand the "Telco Lines -> Gateway T1". If the Gateway T1 plugs into only 1 FXS port, is that FXS port only capable of handling 2 channels, i.e., one call? Thanks in Advnace, Nick. On Thu, Nov 3, 2011 at 9:24 PM, James Sharp <james at fivecats.org> wrote:> On 11/03/2011 09:16 PM, Nick Khamis wrote: >> >> Hello James, >> >> Thank you so much for your response. We just purchased an AudioCodes >> MP124 for this job. And setting >> up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the >> Telco here in Toronto. As for other >> Telcos around the world, for example Bell South in the states, is it >> possible to have them route a block of >> Florida phone numbers to our FXS port here in Canada, or do we have to >> have a T1 gateway + SIP Proxy in Florida, >> routing the calls to our setup in Toronto and vice versa? > > Routing Florida numbers up to Canada would get you charged LD per minute > fees. ?You can go with a provider like Level 3 or Global Crossing and they > can hand you a T1 circuit that has DIDs from many different areas in the US. > > -- > _____________________________________________________________________ > -- 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 >
One FXS port can only handle one call. A PRI T1 gateway can handle 23 call channels. A single T1 Data line with SIP can handle about 18 call channels running G711, 37 channels running g729 Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---------------------------------------- From: "Nick Khamis" <symack at gmail.com> Sent: Thursday, November 03, 2011 10:09 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] DID from Direct from Telco Fair enough, In regards to the the diagram discussed earlier: Telco Lines -> Gateway T1 -> SIP Proxy -> Media Servers -> Customer I understand that a T1 Gateway that has 480 channels, can handle up to 240 calls. That is more than enough for the "Gateway T1 -> SIP Proxy" part of the diagram. I just want to make terribly sure I understand the "Telco Lines -> Gateway T1". If the Gateway T1 plugs into only 1 FXS port, is that FXS port only capable of handling 2 channels, i.e., one call? Thanks in Advnace, Nick. On Thu, Nov 3, 2011 at 9:24 PM, James Sharp <james at fivecats.org> wrote:> On 11/03/2011 09:16 PM, Nick Khamis wrote:>>>> Hello James,>>>> Thank you so much for your response. We just purchased an AudioCodes>> MP124 for this job. And setting>> up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the>> Telco here in Toronto. As for other>> Telcos around the world, for example Bell South in the states, is it>> possible to have them route a block of>> Florida phone numbers to our FXS port here in Canada, or do we have to>> have a T1 gateway + SIP Proxy in Florida,>> routing the calls to our setup in Toronto and vice versa?>> Routing Florida numbers up to Canada would get you charged LD per minute> fees. You can go with a provider like Level 3 or Global Crossing andthey> can hand you a T1 circuit that has DIDs from many different areas in theUS.>> --> _____________________________________________________________________> -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111103/b0521db3/attachment.htm>
I realized there was an error in my last post. I meant analog gateway plugged into and FXO port. DIDs must start somwhere. And I am under the impression that the telcos are the one that have control over that? Therefore, we would first need an analog gateway plugged into an FXO, before being able to go through the T1s and media servers? Your insight is greatly appreciated. Nick.
A telco could either give you a analog line like the old phone line which you
have at home with 1 number and 1 line or a T1 which comes from the telcos office
to yours and plugs directly into a digital gateway with 23 lines and lots of
numbers. and no need at all for analog gateways on the way
If you are going to use a T1 you should return the MP124 you have no need for
that
-----Original Message-----
From: Nick Khamis <symack at gmail.com>
Sender: asterisk-users-bounces at lists.digium.com
Date: Fri, 4 Nov 2011 09:07:11
To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at
lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] DID from Direct from Telco
I realized there was an error in my last post. I meant analog gateway
plugged into and FXO port.
DIDs must start somwhere. And I am under the impression that the
telcos are the one that have
control over that? Therefore, we would first need an analog gateway
plugged into an FXO, before
being able to go through the T1s and media servers? Your insight is
greatly appreciated.
Nick.
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What is your target PBX is it Asterisk? If so your best method is to take calls in direct via SIP trunks, but there are PRI and FXO options available as well. You can not use an FXS gatway to plug to the Telco Service lines. SIP Trunk -> Asterisk or Like VOIP compliant PBX.. If your PBX is not SIP complaint here is a method you can use to get SIP into that. SIP Trunk -> SIP to PRI Grateway - PBX with PRI input. If your PBX does not have the PRI option and only analog channel inputs FXO SIP Trunk -> SIP to FXS Gatway - PBX with FXO inputs Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---------------------------------------- From: isrlgb at gmail.com Sent: Friday, November 04, 2011 9:11 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] DID from Direct from Telco A telco could either give you a analog line like the old phone line which you have at home with 1 number and 1 line or a T1 which comes from the telcos office to yours and plugs directly into a digital gateway with 23 lines and lots of numbers. and no need at all for analog gateways on the way If you are going to use a T1 you should return the MP124 you have no need for that -----Original Message----- From: Nick Khamis <symack at gmail.com> Sender: asterisk-users-bounces at lists.digium.com Date: Fri, 4 Nov 2011 09:07:11 To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at lists.digium.com> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] DID from Direct from Telco I realized there was an error in my last post. I meant analog gateway plugged into and FXO port. DIDs must start somwhere. And I am under the impression that the telcos are the one that have control over that? Therefore, we would first need an analog gateway plugged into an FXO, before being able to go through the T1s and media servers? Your insight is greatly appreciated. Nick. -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111104/13657779/attachment.htm>
It might be a good idea for you to describe your application and ask for suggestions. How many concurrent calls do you need to handle? Do you need a few (or many) DIDs (actual phone numbers)? Are the DIDs in a single geographic area, or scattered all over the country(ies)? Is your application inbound-only, or will you be making outbound calls? Or will you be redirecting calls to outside agents? What is there about the SIP providers that you find unsatisfactory? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis Sent: Friday, November 04, 2011 7:47 AM To: isrlgb at gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID from Direct from Telco Thank you guys for your response,>> One FXS port can only handle one call. A PRI T1 gateway can handle 23 >> call channels. A single T1 Data line with SIP can >> handle about 18 >> call channels running G711, 37 channels running g729I just want to make sure that a T1 Gateway (capable of 23 call channels), plugged into an FXS port (capable of one call), is not a bottleneck. I.e., even though our network can handle upto 23 channels, we can only support 1 concurrent call becuase of the single FXS? What I am trying to figure out is what would I need to have the same capabilities as a company offering DIDs. Which mediant, and maybe a nice illustration? Thanks in Advance, Nick. -- _____________________________________________________________________ -- 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
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.
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-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Nick Khamis
Sent: Friday, November 04, 2011 10:33 AM
To: bryantz at 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.
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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 <EWieling at 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-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis > Sent: Friday, November 04, 2011 10:33 AM > To: bryantz at 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 >
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-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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 <EWieling at 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-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick > Khamis > Sent: Friday, November 04, 2011 10:33 AM > To: bryantz at 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
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" <symack at gmail.com> Sent: Friday, November 04, 2011 10:40 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> 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 <EWieling at 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-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis> Sent: Friday, November 04, 2011 10:33 AM> To: bryantz at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111104/eded957d/attachment.htm>
Hello Bryant, Thank you so much for your insight. This is the exactly direction we are headed. Collocating equipment to different regions here in Canada, and performing least-cost routing. Thanks Again, Nick. On Fri, Nov 4, 2011 at 10:57 AM, Bryant Zimmerman <BryantZ at zktech.com> wrote:> 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" <symack at gmail.com> > Sent: Friday, November 04, 2011 10:40 AM > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > 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 <EWieling at 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-bounces at lists.digium.com >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis >> Sent: Friday, November 04, 2011 10:33 AM >> To: bryantz at 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 >