DID from Direct from Telco

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Asterisk Users 15 Comments

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,


15 thoughts on - DID from Direct from Telco

  • 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,


  • 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,


  • The mp124 is a analog gateway and doesn’t support t1’s I think

    A T1 is a digital line which has 24 channels per port which means 24 calls concurrently if you want more channels you need more ports

    DID’s are incoming numbers the telco sends down your trunk(port) you could have thousands of DID’s on 1 T1

    You need a digital gateway for connecting to a T1

    Did you check if your provider will give you a T1 or maybe they could provide you a sip trunk which will save you on the hardware

  • 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


    Bryant Zimmerman (ZK Tech Inc.)

    616-855-1030 Ext. 2003

  • Thank you guys for your response,

    I 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,


  • 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.


  • 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

  • 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

    SIP Trunk -> SIP to FXS Gatway – PBX with FXO inputs


    Bryant Zimmerman (ZK Tech Inc.)

    616-855-1030 Ext. 2003

  • It might be a good idea for you to describe your application and ask for

    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

  • 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

    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,


  • 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.

  • 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?