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I need to terminate VoIP traffic from USA on one location in Europe where I
have 1 E1 (30 ISDN channels) and 512 mbps Internet access. Quality of service must be pretty good. From my understanding I need one Linux server running Asterix and Digium Wildcard E100P. All termination is originated from one location. I need also database of call records so I can see how much traffic is terminated in one period of time. Is Asterix good solution for this job? Shark |
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Shark <> wrote:
> I need to terminate VoIP traffic from USA on one location in Europe where I > have 1 E1 (30 ISDN channels) and 512 mbps Internet access. > > Quality of service must be pretty good. From my understanding I need one > Linux server running Asterix and Digium Wildcard E100P. All termination is > originated from one location. I need also database of call records so I can > see how much traffic is terminated in one period of time. > > Is Asterix good solution for this job? Asterisk will do all that easily. I think your bigger problem is going to be squeezing 30 voice calls into 512 megabits. miguel |
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"Miguel Cruz" <> wrote in message news:s0JMa.7769$n%... > Shark <> wrote: > > I need to terminate VoIP traffic from USA on one location in Europe where I > > have 1 E1 (30 ISDN channels) and 512 mbps Internet access. > > > > Quality of service must be pretty good. From my understanding I need one > > Linux server running Asterix and Digium Wildcard E100P. All termination is > > originated from one location. I need also database of call records so I can > > see how much traffic is terminated in one period of time. > > > > Is Asterix good solution for this job? > > Asterisk will do all that easily. > > I think your bigger problem is going to be squeezing 30 voice calls into > 512 megabits. > > miguel Sorry, this was a mistake in writing, I meant 512 kbps. This bandwith can be upgraded, what is optimal for 30 voice calls? All calls are originating from USA, from regular telephone network, converted by Cicso 5300 and/or Quintum Tenor VoIP gateways. Asterix should terminate this traffic and use standard protocols which this gateways use. Is it ok solution? How strong computer must be to decompress 30 voice chanells? Thank you a lot in advance! |
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For only 30 channels of voice you wont need anything beefy beyond a 600 mhz
PC, it depends on what the big picture is, what codecs do you plan to use, what protocols? Asterisk can handle a multitide of codecs and protocols but the more I know about your project the better I can respond with something that will scale. Outside of what you've mentioned already, is there anything else you want it to do? Keep in mind Asterisk is: Heterogeneous Voice over IP gateway (MGCP, SIP, H.323,IAX) Private Branch eXchange (PBX) Custom Interactive Voice Response (IVR) server Softswitch Conferencing server Number translation Calling card application Predictive dialer Call queuing with remote agents Remote offices for existing PBX Voicemail server And can be a complete telephone solution with support, a custom GUI and Billing Platform from The NuFone Network. Greg Merriweather Systems Engineer The NuFone Network "Shark" <> wrote in message news:be14p5$1vrf$... > > "Miguel Cruz" <> wrote in message > news:s0JMa.7769$n%... > > Shark <> wrote: > > > I need to terminate VoIP traffic from USA on one location in Europe > where I > > > have 1 E1 (30 ISDN channels) and 512 mbps Internet access. > > > > > > Quality of service must be pretty good. From my understanding I need one > > > Linux server running Asterix and Digium Wildcard E100P. All termination > is > > > originated from one location. I need also database of call records so I > can > > > see how much traffic is terminated in one period of time. > > > > > > Is Asterix good solution for this job? > > > > Asterisk will do all that easily. > > > > I think your bigger problem is going to be squeezing 30 voice calls into > > 512 megabits. > > > > miguel > > Sorry, this was a mistake in writing, I meant 512 kbps. > > This bandwith can be upgraded, what is optimal for 30 voice calls? > > All calls are originating from USA, from regular telephone network, > converted by Cicso 5300 and/or Quintum Tenor VoIP gateways. Asterix should > terminate this traffic and use standard protocols which this gateways use. > Is it ok solution? > > How strong computer must be to decompress 30 voice chanells? > > Thank you a lot in advance! > > > |
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