<div>Hi,</div>
<div> </div>
<div>I would like to read your comments for the following setup:</div>
<div> </div>
<div>Building A:</div>
<div>3 voice E1 incoming to a quad redfone fonebridge (TDMoE)</div>
<div>The fonebridge goes to a port in a 24 port gigabit switch</div>
<div>in the gigabit switch VLAN1 is for the fonebridge and the first gigabit NIC on a dual NIC server</div>
<div>in the gigabit switch VLAN2 is for the second gigabit NIC card on the server and eleven 10/100 switches with 250 SIP phone users running g711 codec (24 phones per 10/100 switch,each switch is 24port)</div>
<div>Building A and Building B are connected over a 10Mbits fiber link.</div>
<div>Numeric Extensions at building A are 1xxx</div>
<div> </div>
<div>Building B:</div>
<div>same config E1/switch/users as building A</div>
<div>
<div>Building A and Building B are connected over a 10Mbits fiber link.</div></div>
<div>
<div>Numeric Extensions at building B are 2xxx</div>
<div> </div>
<div>The asterisk servers at each side will talk IAX2 between each other for building-to-building call transfers.</div>
<div> </div>
<div>Suggested machine:</div></div>
<div>Im considering a Dell PowerEdge 9G 1950, Dual Xeon 3.20Ghz, 1066 FSB, 4GB ram. two 73GB SAS 15k RPMs hard disk and dual gbit network card.</div>
<div> </div>
<div>Asterisk Features:</div>
<div>Music on hold</div>
<div>call transfer</div>
<div>call waiting (but only on executive phones, around 20)</div>
<div>voicemail</div>
<div>a small queue (about 10 persons)</div>
<div>and a simple IVR (play prompts for department selection, transfer according to selection).</div>
<div>No call recording requested at this time.</div>
<div> </div>
<div>Operating System:</div>
<div>Centos 4.3</div>
<div> </div>
<div>Codecs: G711 for the SIP to asterisk and IAX for server to server transfers. If IAX is not recommended, please advice.</div>
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<div>Notes:</div>
<div>a- Is is expected to have the 250 SIP users talking either to each other and/or to the other building and/or to the fonebridge E1s.</div>
<div>b- I know that for SIP-to-ZAP a calculation of 30Mhz per voice channel is a rule of thumb, but i also read somewhere that the same calculation does not apply when doing "Pure IP, no SIP/ZAP and pure g711 implementations"
</div>
<div>I'm in that category.</div>
<div>c-Just for the record, what if I change to g729?</div>
<div>d- It is expected to have 80% of the calls over the E1 being incoming from the PSTN and the other 20% ar the SIP users calls to the PSTN</div>
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<div>Is is also expected to have one 24 port Rhino FXS channel banks connected to the 4th port of the fonebridge. Is used, it will add another 24 users to the setup.</div>
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<div>Thanks in advance. Your comments are welcomed.</div>
<div><br> <br>------------------------------------------------------------<br>Erick Perez<br>Panama Sistemas<br>Integradores de Telefonia IP y Soluciones Para Centros de Datos<br>Panama, Republica de Panama<br>Cel Panama. +(507) 6694-4780
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