Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as this will be some kind of a "hosted pbx". Given he finds a provider wich has this much SIP capacity and IP bandwith and no codec conversion is needed - do you think this is possible with pure asterisk on a decent system? Is there anything I shoudl watch out for? Your help is much appreciated! Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090324/92f32bbb/attachment.htm
First Issue to be addressed is how many simultaneous calls and bandwidth availability. Number of "lines" (numbers) is not a limitation in it self unless they are all in use. Cary Fitch _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Christian Victor Sent: Tuesday, March 24, 2009 10:10 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] SIP trunk with > 250 lines Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as this will be some kind of a "hosted pbx". Given he finds a provider wich has this much SIP capacity and IP bandwith and no codec conversion is needed - do you think this is possible with pure asterisk on a decent system? Is there anything I shoudl watch out for? Your help is much appreciated! Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090324/81014714/attachment-0001.htm
2009/3/24 Christian Victor <christian at victormedia.de>> Hi! > > A customer of mine wants to connect an asterisk system with 240 to 480 > lines to a PSTN switch. To save the costs for E1 cards and the corresponding > E1 mainlines he wants to connect the system to the switch by a SIP trunk. > > Phones will be connected to the server through the same SIP trunk as this > will be some kind of a "hosted pbx". > > Given he finds a provider wich has this much SIP capacity and IP bandwith > and no codec conversion is needed - do you think this is possible with pure > asterisk on a decent system? Is there anything I shoudl watch out for? > > Your help is much appreciated! > > Chris > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >If the switch is fine why not ? But i wander why kind if switch is that 240-480 fxo ? ;) Sounds like a big overkill. And i dont see a problem with asterisk, if not too much transcoding involved and with the right hardware. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090324/3d38ccd2/attachment.htm
2009/3/24 Cary Fitch <caryf at usawide.net>> First Issue to be addressed is how many simultaneous calls and bandwidth > availability. > > Number of ?lines? (numbers) is not a limitation in it self unless they are > all in use. >Sorry for being a bit unclear in this point. What I meant was 240 to 480 concurrent active calls. Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090324/adb0cfaf/attachment.htm
I use a Dell with the 1Gb Ethernet on board, but had to clock it down to 100 Mhz because * has an issue with Dell on board Ethernet. _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Christian Victor Sent: Tuesday, March 24, 2009 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP trunk with > 250 lines 2009/3/24 Danny Nicholas <danny at debsinc.com> Here are a few "look outs"; Using conference rooms will increase your bandwidth requirements. On board Network controllers will affect performance in this "high-use" scenario. 250 simultaneous calls will use about 7.5Mb of bandwidth depending on the codec(s) you use. I think we will use G.711a to prevent transcoding. So I calculate with 25 to 50 Mbit. I hope a dedicated 1GB Ethernet (although on board) will do it. MeetMe will not be used. Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090324/1df18d61/attachment.htm
Hi, We have a customer who used a strong quad-core Xeon box to convert up to 800 simultneous calls from SIP to IAX and trunk them to another box. So your requirement doesn't look like a big problem. Steve On 3/24/09, Christian Victor <christian at victormedia.de> wrote:> Hi! > > A customer of mine wants to connect an asterisk system with 240 to 480 lines > to a PSTN switch. To save the costs for E1 cards and the corresponding E1 > mainlines he wants to connect the system to the switch by a SIP trunk. > > Phones will be connected to the server through the same SIP trunk as this > will be some kind of a "hosted pbx". > > Given he finds a provider wich has this much SIP capacity and IP bandwith > and no codec conversion is needed - do you think this is possible with pure > asterisk on a decent system? Is there anything I shoudl watch out for? > > Your help is much appreciated! > > Chris >-- Sent from my mobile device