Erick Perez
2006-Aug-29 13:00 UTC
[asterisk-users] CPU configuration for 250 calls SIP to SIP to IAX and fonebridge and two asterisk servers
Hi, I would like to read your comments for the following setup: Building A: 3 voice E1 incoming to a quad redfone fonebridge (TDMoE) The fonebridge goes to a port in a 24 port gigabit switch in the gigabit switch VLAN1 is for the fonebridge and the first gigabit NIC on a dual NIC server 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) Building A and Building B are connected over a 10Mbits fiber link. Numeric Extensions at building A are 1xxx Building B: same config E1/switch/users as building A Building A and Building B are connected over a 10Mbits fiber link. Numeric Extensions at building B are 2xxx The asterisk servers at each side will talk IAX2 between each other for building-to-building call transfers. Suggested machine: 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. Asterisk Features: Music on hold call transfer call waiting (but only on executive phones, around 20) voicemail a small queue (about 10 persons) and a simple IVR (play prompts for department selection, transfer according to selection). No call recording requested at this time. Operating System: Centos 4.3 Codecs: G711 for the SIP to asterisk and IAX for server to server transfers. If IAX is not recommended, please advice. Notes: 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. 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" I'm in that category. c-Just for the record, what if I change to g729? 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 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. Thanks in advance. Your comments are welcomed. ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060829/d8e88a7d/attachment.htm