Dear all i'm planning a migration to asterisk for a high volume IVR service (from 1000 to 1500 concurrent call) The IVR service is based only on DTMF tones so the features required is - play feature - dtmf detection Asterisk will receive calls via VOIP (SIP with g711 codec) The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in the same codec of the received call) and some dtmf detections. How many simultaneous call can i handle per server? each server will have: 4 core 3.0 Ghz 4 GB of RAM I need an aproximate sizing: 0-100 calls per server ? 100-200 calls per server ? 200-300 calls per server ? 300-400 calls per server? 400-500 calls per server? Thanks to all in advance -- /*************/ nik600 http://www.kumbe.it
Leandro Dardini
2013-Mar-08 12:17 UTC
[asterisk-users] asterisk sizing for play and dtmf detection
2013/3/8 nik600 <nik600 at gmail.com>> Dear all > > i'm planning a migration to asterisk for a high volume IVR service > (from 1000 to 1500 concurrent call) > > The IVR service is based only on DTMF tones so the features required is > > - play feature > - dtmf detection > > Asterisk will receive calls via VOIP (SIP with g711 codec) > > The IVR service wil be a static service based on Asterisk dialplan > with some prompt (from 0 to 5, play of files in the same codec of the > received call) and some dtmf detections. > > How many simultaneous call can i handle per server? each server will have: > > 4 core 3.0 Ghz > 4 GB of RAM > > I need an aproximate sizing: > > 0-100 calls per server ? > 100-200 calls per server ? > 200-300 calls per server ? > 300-400 calls per server? > 400-500 calls per server? > > Thanks to all in advance > > -- > /*************/ > nik600 > http://www.kumbe.it > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >The bigger server I have has 150 max channels during peak hours and has a max load of 0.5 with 24 cores. When I was using a 4 cores server with the same number of channels, I get a load of 3 ... so the load x core relation is valid. I think it will be good to have a load not over 4 for a 4 core server, so you can have at least 200 active channels on the server. If you accept more load, then you can get more channels. Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130308/68df933f/attachment.htm>
Mitul Limbani
2013-Mar-08 12:25 UTC
[asterisk-users] asterisk sizing for play and dtmf detection
If you accept calls on.g711 and static ivr dialplan you should be able to do around 300-400 concurrent on the box config that you provided. And If you pay some expert consultant, he may be to fine tune it to be able to handle 500 concurrent as well. Which version of asterisk are you planning to use? Any DB integration layer inside IVR? Mitul Limbani On Mar 8, 2013 5:20 PM, "nik600" <nik600 at gmail.com> wrote:> Dear all > > i'm planning a migration to asterisk for a high volume IVR service > (from 1000 to 1500 concurrent call) > > The IVR service is based only on DTMF tones so the features required is > > - play feature > - dtmf detection > > Asterisk will receive calls via VOIP (SIP with g711 codec) > > The IVR service wil be a static service based on Asterisk dialplan > with some prompt (from 0 to 5, play of files in the same codec of the > received call) and some dtmf detections. > > How many simultaneous call can i handle per server? each server will have: > > 4 core 3.0 Ghz > 4 GB of RAM > > I need an aproximate sizing: > > 0-100 calls per server ? > 100-200 calls per server ? > 200-300 calls per server ? > 300-400 calls per server? > 400-500 calls per server? > > Thanks to all in advance > > -- > /*************/ > nik600 > http://www.kumbe.it > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130308/59293905/attachment.htm>
Apparently Analagous Threads
- wrond DTMF detection on Zap channel
- put some IVR into a queue after the call queuing
- server sizing for ~ 200 simultaneous call
- problem with DTMF detection on calls created with Originate AMI command
- what can we do with lost voice packet on a congestioned VPN?