Re: [asterisk-users] asterisk with 1000 extensions
-Original Message- From: Carlos Alvarez car...@televolve.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk with 1000 extensions Date: Thu, 7 Mar 2013 09:30:31 -0700 On Thu, Mar 7, 2013 at 1:44 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flat Secondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Grandstream makes some inexpensive phones that are still very good. Cheapest hasn't been defined yet. What's the budget? Is there existing networking at these locations? Will you need switches? PoE? -Original Message- I think Carlos said it properly. Anything related to asterisk is insignificant compared to the rest. I dare to say, that the requirements if for 1000 people to communicate between themselves. So why SIP-phones? Why VOIP at all? Look at it a bit broader: network, maintenance (people), power, ... -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
2013/3/7 Steve Edwards asterisk@sedwards.com Please don't top-post. On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) I bet it is a school assignment ... home work or the way you like to call them. However I have a box with 972 peers, no reinvite (but no transcoding), average usage of conference call and other audio mix feature, reaching a max of 60 CPS and an average of 150 channels without problems. The cpu is a double Intel(R) Xeon(R) CPU E5-2630 0 @ 2.30GHz, but it works fine even on the old hardware, a double Intel(R) Xeon(R) CPU 5150 @ 2.66GHz Leandro -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
2013/3/7 Steve Edwards asterisk@sedwards.com Please don't top-post. On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) I bet it is a school assignment ... home work or the way you like to call them. However I have a box with 972 peers, no reinvite (but no transcoding), average usage of conference call and other audio mix feature, reaching a max of 60 CPS and an average of 150 channels without problems. The cpu is a double Intel(R) Xeon(R) CPU E5-2630 0 @ 2.30GHz, but it works fine even on the old hardware, a double Intel(R) Xeon(R) CPU 5150 @ 2.66GHz Leandro, This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flatSecondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Thanks,Kamlesh -- _ -- 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 -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
On 7/03/2013, at 9:29 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flat Secondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Your costs will be in the handsets. Yealink make good cheap phones, you need to find a supplier who can do you a great deal on 1000 phones http://www.yealink.com/product_info.aspx?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147 But I am not sure why you cant use analogue phones and SIP channel banks such as grandstream or USB ones such as Xorcom. The per line cost will come down and you only need telephony grade cabling to the premise. You can get $10 phones which limit the desire of people to walk off with them The server and setup will cost nothing compared to the handsets Thanks, Kamlesh -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
2013/3/7 Duncan Turnbull dun...@e-simple.co.nz On 7/03/2013, at 9:29 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flat Secondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Your costs will be in the handsets. Yealink make good cheap phones, you need to find a supplier who can do you a great deal on 1000 phones http://www.yealink.com/product_info.aspx?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147 But I am not sure why you cant use analogue phones and SIP channel banks such as grandstream or USB ones such as Xorcom. The per line cost will come down and you only need telephony grade cabling to the premise. You can get $10 phones which limit the desire of people to walk off with them The server and setup will cost nothing compared to the handsets Thanks, Kamlesh Sorry, but it is not the first time we help little boys to make homework, it seems asterisk course are common in India and it is easier to cheat than to apply. If you are really trying to serve 1000 phones, beside the usage of SIP or analogue phones via channel banks, I think it will be better to not handle all the load on a single server, but to spread the phone among multiple servers. The best will be to have multiple asterisks working together using realtime extensions. It is not difficult to make. Leandro -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
On 7/3/13 6:50 am, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. I would caution against that approach. Analogue to Digital conversions often seem to have 'problems' - mostly related to hangup detection and/or echo. If you really do want to use analogue phones, then use a good quality channel bank to bring the analogue extensions into Asterisk, not low-end ATAs. You also have to consider the value of your time. There's little point shaving a few pounds (or dollars, or euros) from the hardware cost if it's going to double the configuration time. And using 'cheap' components will add to your ongoing support burden for the system. Cheap != good value for money. Personally, I'd consider using something like the Snom 710. They aren't the cheapest SIP phones by any means, but they do have a very good remote provisioning and configuration system, which will substantially reduce the work you need to do in configuring handsets. If your budget won't stretch to the Snom units, the Yealink range as suggested by another poster might be worth looking at. I believe their cheapest (is it the T18?) SIP endpoint can be had for around 35GBP - I don't know what pricing is like in your local currency of course. I believe Yealink do also have a fairly reasonable remote provisioning system, but unlike the Snom system, I can't claim to have used it in anger. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
From: Chris Bagnall aster...@lists.minotaur.cc To: asterisk-users@lists.digium.com, Date: 03/07/2013 06:43 AM Subject:Re: [asterisk-users] asterisk with 1000 extensions Sent by:asterisk-users-boun...@lists.digium.com On 7/3/13 6:50 am, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Cheap != good value for money. I am going to go with Chris on this. Don't look for the absolute cheapest, look for the absolute best value, balancing the cost of the end units versus the cost spent to support them. We use Polycom phones and while they are not the cheapest, I never have to mess with them other than to update configs to add in some new feature I am working on. Rock solid, even after 5 years and many firmware upgrades. -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
On Thu, Mar 7, 2013 at 1:44 AM, Duncan Turnbull dun...@e-simple.co.nzwrote: This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flat Secondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Grandstream makes some inexpensive phones that are still very good. Cheapest hasn't been defined yet. What's the budget? Is there existing networking at these locations? Will you need switches? PoE? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
On Thu, 7 Mar 2013, Kamlesh Kumar wrote: We need to setup asterisk server for 1000 extensions and in this setup only extension to extension dialling is required (without call recording and voicemail), like intercom calling. Please let us know what can be the best economic solution/setup for this. The number of extensions is not the key factor. The number of simultaneous calls is. What technology? SIP? Dahdi? If all you are going to do is call from endpoint to endpoint, maybe something like Kamailio or OpenSIPS is appropriate. If Asterisk is not handling the media, probably any old crappy computer can handle the call setup/call teardown load. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
Technology is SIP and asterisk is not handling the media, what is cheapest solution to be used for SIP client. Thanks,Kamlesh Date: Wed, 6 Mar 2013 20:43:52 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk with 1000 extensions On Thu, 7 Mar 2013, Kamlesh Kumar wrote: We need to setup asterisk server for 1000 extensions and in this setup only extension to extension dialling is required (without call recording and voicemail), like intercom calling. Please let us know what can be the best economic solution/setup for this. The number of extensions is not the key factor. The number of simultaneous calls is. What technology? SIP? Dahdi? If all you are going to do is call from endpoint to endpoint, maybe something like Kamailio or OpenSIPS is appropriate. If Asterisk is not handling the media, probably any old crappy computer can handle the call setup/call teardown load. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
On Thu, 7 Mar 2013, Kamlesh Kumar wrote: Technology is SIP and asterisk is not handling the media, what is cheapest solution to be used for SIP client. Client? How about a free SIP softphone? Server? How many calls per second? How many simultaneous calls? Any half-way recent box should do. An Atom, i3, etc. Reliability and redundancy are going to be important unless you want 1,000 people calling you :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
softphone is not going to be used in this setup. Hardphone is required. Around 60-70 simultaneous calls would be required. Thanks,Kamlesh Date: Wed, 6 Mar 2013 21:15:51 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk with 1000 extensions On Thu, 7 Mar 2013, Kamlesh Kumar wrote: Technology is SIP and asterisk is not handling the media, what is cheapest solution to be used for SIP client. Client? How about a free SIP softphone? Server? How many calls per second? How many simultaneous calls? Any half-way recent box should do. An Atom, i3, etc. Reliability and redundancy are going to be important unless you want 1,000 people calling you :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
Please don't top-post. On Thu, 7 Mar 2013, Kamlesh Kumar wrote: softphone is not going to be used in this setup. Hardphone is required. Around 60-70 simultaneous calls would be required. OK. So figure on about 6 UDP packets, about 3.5 KB per call. Not a big deal. I'd look for a reliable system and build 2 so you can swap between them as needed. Going the full redundant, heartbeat kind of setup may be more trouble than it is worth depending on how tolerant your users are to the very occasional outage. A couple of years ago, I bought a used Supermicro server with a 3.2 Ghz P4 off Ebay for $150 including shipping. Earlier this week, I updated the OS and rebooted it. The uptime was 574 days. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
Server side installation with recent hardware is fine, we can build two parallel system for redundancy. We are more concern with the cost of SIP client (hardphone). What are the various ways to make this setup functional with low cost for SIP clients. Thanks,Kamlesh On Thu, 7 Mar 2013, Kamlesh Kumar wrote: softphone is not going to be used in this setup. Hardphone is required. Around 60-70 simultaneous calls would be required. OK. So figure on about 6 UDP packets, about 3.5 KB per call. Not a big deal. I'd look for a reliable system and build 2 so you can swap between them as needed. Going the full redundant, heartbeat kind of setup may be more trouble than it is worth depending on how tolerant your users are to the very occasional outage. A couple of years ago, I bought a used Supermicro server with a 3.2 Ghz P4 off Ebay for $150 including shipping. Earlier this week, I updated the OS and rebooted it. The uptime was 574 days. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] asterisk with 1000 extensions
Please don't top-post. On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
