Thanks!<br><br>That is some good info, I will check with them.<br><br><br><div class="gmail_quote">On Thu, Apr 15, 2010 at 1:49 AM, Vahan Yerkanian <span dir="ltr"><<a href="mailto:vahan@arminco.com">vahan@arminco.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000"><div class="im">
On 4/15/10 1:26 AM, Tonty T wrote:
<blockquote type="cite">That's is all the overhead I am trying to avoid. What I
need is a DID with unlimited channel, but they do not offer DIDs in
that country. I wanted to know for example when I get a DID from lets
say Vitelity, with unlimited channel, what are they using to forward
the calls via SIP or IAX to my server? If I knew the details of the
process, I could probably tell them to used this method and route the
short code to me via SIP. And if it requires hardware I could invest
in it myself and have them host it.
<div><br>
</div>
</blockquote></div>
If their switch doesn't support SIP or doesn't have SIP module
installed, there isn't much you can do to get traffic in pure SIP form.
Ask them if they can and willing to serve you the traffic via multiple
E3 or even better, STM fiber links. STM over fiber is the cheapest way
to transport that much channels by means of cabling - you just need 2
strands for TX/RX or even 1 strand if you go with WDM. However the
carrier crade hardware for it is *very expensive*. On your side you
demux STM link(s) into E3/E1s using expensive carrier grade equipment
like Cisco's $25k+ (used) STM cards for Cisco 7500 and up models or if
you're smart enough to know where to dig, dirt cheap (~$2K for STM-1 to
24E1) Taiwanese/Chinese media converters.<br>
<br>
Oh and yes, this isn't a task for a single Asterisk server. The most
I've seen a single box capable of is 16 E1s (2 x 8E1 cards) in a single
chassis doing only G711a to SIP conversion.<br>
<br>
HTH,<br><font color="#888888">
Vahan</font><div><div></div><div class="h5"><br>
<blockquote type="cite">
<div><br>
<div class="gmail_quote">On Wed, Apr 14, 2010 at 2:46 PM, Jeff Brower
<span dir="ltr"><<a href="mailto:jbrower@signalogic.com" target="_blank">jbrower@signalogic.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div>> On Wed, Apr 14, 2010 at 10:33 AM, Tonty T <<a href="mailto:tonty2@gmail.com" target="_blank">tonty2@gmail.com</a>>
wrote:<br>
><br>
>> This is a solution they proposed, using GSM gateways, but it
wont let me<br>
>> handle 1000 simultaneous calls, the other option was using an
E1 but the<br>
>> cost would be too much to deploy 35 E1s to support that many
calls. There<br>
>> might be a better way of doing it.<br>
>><br>
>><br>
> If you are planning on having 1000 simultaneous calls, you're
going to be<br>
> looking at a hefty price tag one way or the other. Things to
consider - if<br>
> you're going to have 1000 concurrent calls going out over VoIP
trunks (SIP /<br>
> IAX / whatever), you need to have enough bandwidth to comfortably
handle<br>
> that many calls (each g729 is 8Kb/s bandwidth (but you need to pay
a license<br>
> fee for each channel of g729), each g711alaw is 64Kb/s, etc). That
amount<br>
> of bandwidth won't be cheap, plus the cost of the ITSP giving your
1000<br>
> concurrent channels to call on. On the other hand, if you have a
bank of<br>
> E1's, which support (I think) at max 30 concurrent voice channels,
you'd<br>
> need 34 available E1 spans. I'm not sure if you can get 34 spans
working in<br>
> a single asterisk server (there was some discussion about this
recently on<br>
> this list), and you'd have the cost of 34 E1 spans as well.<br>
<br>
</div>
</div>
All good points. It might be worth mentioning that including
IP/UDP/RTP packet overhead, actual bandwidth is 40 kbps<br>
for G729 and 96 kbps for G711.<br>
<br>
-Jeff<br>
<font color="#888888"><br>
<br>
<br>
--<br>
</font>
<div>
<div>_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a>
--<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
<br>
</div></div></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br>