<div dir="ltr"><div>Kristian!! I guess it takes a real juicy technical conundrum to bring you out of the woodworks! :) <br><br></div>-MC<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jun 18, 2013 at 11:10 AM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kris@kriskinc.com" target="_blank">kris@kriskinc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Assuming this is serious I would reconsider your numbers, resources,<br>
and overall approach.<br>
<br>
1000 cps results in a 6.5 second call time across 10,000 simultaneous<br>
channels assuming 100% connect rates (which doesn't happen). 20,000<br>
simultaneous channels would be 1000 cps with 16.5 second call times...<br>
<br>
Or your CPS rate is absurdly high because this is dialer traffic with<br>
very poor quality number lists. If that's the case you won't be able<br>
to find anyone to take this traffic from you.<br>
<br>
Dual Sangoma octal cards per chassis results in 384 TDM (max) ports<br>
per machine. That's 27 machines with some crazy load balancing in<br>
front of it. Rough back of the envelope math shows at least $165K in<br>
hardware expense...<br>
<br>
OR you can try to get 10,000 ports of SIP directly from someone, but<br>
again considering these metrics you'd be hard pressed to find anyone<br>
(TDM or SIP) willing to accept these rates.<br>
<br>
If this is a real project you'd be well served consulting with some of<br>
the professionals on this list (and others) that have experience with<br>
projects of this scale. You can start with e-mailing<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a>.<br>
<br>
On Mon, Jun 17, 2013 at 10:15 PM, Andre Demattia <<a href="mailto:andretodd@verizon.net">andretodd@verizon.net</a>> wrote:<br>
> Ds3 or T1. Anything to terminate sip traffic.<br>
> Pretty much something that I can be my own sip provider. I'm not sure what<br>
> equipment to get. I need 1000's if ports and 1000 cps.<br>
> ________________________________<br>
> From: Guillermo Ruiz Camauer<br>
> Sent: 6/17/2013 9:56 PM<br>
><br>
> To: FreeSWITCH Users Help<br>
> Subject: Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec<br>
><br>
> By dial tone do you mean an analog line?<br>
><br>
> Guillermo<br>
><br>
> Sent from my iPhone<br>
><br>
> On 17/06/2013, at 22:51, Ken Rice <<a href="mailto:krice@freeswitch.org">krice@freeswitch.org</a>> wrote:<br>
><br>
> You mean something to gate SIP to the traditional TDM PSTN? (like T1s or<br>
> E1s?) if so then you are looking for something like an A104DE (4 port T1<br>
> card) or something like an Adtran 900 series gateway if you want a hardware<br>
> device<br>
><br>
><br>
> On 6/17/13 8:40 PM, "Andre Demattia" <<a href="mailto:andretodd@verizon.net">andretodd@verizon.net</a>> wrote:<br>
><br>
> Hmm I was looking for something to turn sip traffic to dial tone. I guess<br>
> this isn't what I need.<br>
> ________________________________<br>
> From: Guillermo Ruiz Camauer <mailto:<a href="mailto:grcamauer@gmail.com">grcamauer@gmail.com</a>><br>
> Sent: ?6/?17/?2013 9:07 PM<br>
> To: FreeSWITCH Users Help <mailto:<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
> Subject: Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec<br>
><br>
> The simple andwer is NO. The Sangoma D500 is a transcoding card. It<br>
> converts an RTP stream in g729 to PCMU for example. It does absolutely<br>
> nothing with SIP signaling. It is recommended for high traffic systems or<br>
> for a large number of G729 licenses. Note that the D500 can be shared among<br>
> several servers.<br>
> There are smaller cards like the D100 which is the one I have which run up<br>
> to 120 simultaneous transcoding streams.<br>
> A similar card is the Digium TCE400B.<br>
><br>
> Guillermo<br>
><br>
> Sent from my iPhone<br>
><br>
> On 17/06/2013, at 20:07, Andre <<a href="mailto:andretodd@verizon.net">andretodd@verizon.net</a>> wrote:<br>
><br>
> HI, has anyone used Sangoma D500 with freeswitch and if so how does it work?<br>
><br>
> Does this card connected to a Gig E become a SIP Trunk?<br>
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<br>
<br>
--<br>
Kristian Kielhofner<br>
<br>
_________________________________________________________________________<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
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</div>