mattf
2005-May-23 17:40 UTC
[Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues
For an inbound call center with 4 T1s and 30-50 agents on you would do just fine with a single, one-processor machine. We have handled more than this on a single P4 server although we use astGUIclient instead of Asterisk queues, but the load is very similar. I would recommend a Sangoma Quad T1 card because they are about 30% more efficient than Digium T1 cards. When you say that you need to scale to 100s of consecutive calls, is that closer to 200 or 900? and what timeframe is that planned for? We have a distributed in/outbound call center environment across 4 geographic locations with over 20 T1s connected so it is possible for Asterisk to handle over 1000 consecutive calls across the system if you design it right. One of the reasons we don't use Asterisk queues, other than the difficulty in customizing the code to work with the ManagerAPI and client apps, is that it was hard to scale across multiple servers. That's why we use the astGUIclient suite which is more customizable and scalable across multiple servers, although (and it pains me to say this because we developed it) it is not as easy to install and setup than just creating an Asterisk queue. We use a combination of SIP, IAX and Zap client phones depending on the system and the user and yes 711 is always best to use when you can. And if you have many remote phones using a codec like GSM it may actually be better to have a dedicated machine doing nothing but the transcoding from GSM->711 and then just using IAX or a crossover Zap T1 to the inbound server to reduce processor load. In any case it is always advisable to have a backup server that is fully ready to jump in production with a minimum of reconfiguration. A couple more questions, will you do much recording? and what kind of disks do you plan on using? Hope this helps, MATT--- -----Original Message----- From: Warren Smith [mailto:warren@serverplus.com] Sent: Monday, May 23, 2005 8:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inbound call center - reliability \ scalability with queues We are wanting to move off of our legacy inter-tel phoneswitch and move to VoIP and asterisk. We are looking for a new PBX because the inter-tel switch is too difficult to integrate our existing (and new) software into. We are a technical support center. All our calls currently come in on toll free numbers via T1's, and there are 3 of them. I want to use a media gateway to convert the T1's into SIP VOIP (I want reliable hardware for the gateway), and use asterisk as the PBX having all incoming and outgoing channels as SIP. Almost all dialplans will be using Queues, and there will likely be no more than 10 queues, with (currently) about 80 incoming toll-free numbers. There are approximatley 30 agents, but as of right now there are no more than 15 agents logged in at a time. We need to be able to support 60-70 simultaneous calls initially and we have to be able to do this reliably. We also need to be able to scale into the 100's of simultanous calls range. What would be the best option, to have 2 powerful machines (dual powersupply, ) with one as a hotswappable backup or have multiple machines with a sort of load balancer setup? Having multiple machines could possibly cost less, but I'm not sure how the queues and agents would be managed across multiple machines. I.e. how would the agent 'login' to each asterisk machine so that the calls could be handed to it, and how would the calls get handed to an available agent by 4 seperate asterisk machines? I've read through the wiki, but I'm not sure how much overhead queues would put into the system. I want to have all the codecs the same, so the asterisk machines doesn't do any transcoding, and have all channels as SIP. There will be music on hold. Would a dual 2.8 ghz xeon in this config be able to handle 80 simultaneous incoming calls? Would using the 711 codec make a difference in available processing power? I'm sorry if this has been answered a million times already, I just didn't see many configs close to what we're trying to do to compare to. Thanks for any input you may have.
Warren Smith
2005-May-24 09:37 UTC
[Asterisk-Users] Inbound call center - reliability \ scalability with queues
I would say that we would need to be able to scale to the 200+ consecutive call range in the near future (6 months), and hopefully to the 500+ within the next two years. We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K rpm SCSI). We also are planning on recording all queue calls that are answered, and possibly outbound calls made by support agents since it has proven to be extremely helpful when it comes to training. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mattf Sent: Monday, May 23, 2005 6:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Inbound call center - reliability \ scalability with queues For an inbound call center with 4 T1s and 30-50 agents on you would do just fine with a single, one-processor machine. We have handled more than this on a single P4 server although we use astGUIclient instead of Asterisk queues, but the load is very similar. I would recommend a Sangoma Quad T1 card because they are about 30% more efficient than Digium T1 cards. When you say that you need to scale to 100s of consecutive calls, is that closer to 200 or 900? and what timeframe is that planned for? We have a distributed in/outbound call center environment across 4 geographic locations with over 20 T1s connected so it is possible for Asterisk to handle over 1000 consecutive calls across the system if you design it right. One of the reasons we don't use Asterisk queues, other than the difficulty in customizing the code to work with the ManagerAPI and client apps, is that it was hard to scale across multiple servers. That's why we use the astGUIclient suite which is more customizable and scalable across multiple servers, although (and it pains me to say this because we developed it) it is not as easy to install and setup than just creating an Asterisk queue. We use a combination of SIP, IAX and Zap client phones depending on the system and the user and yes 711 is always best to use when you can. And if you have many remote phones using a codec like GSM it may actually be better to have a dedicated machine doing nothing but the transcoding from GSM->711 and then just using IAX or a crossover Zap T1 to the inbound server to reduce processor load. In any case it is always advisable to have a backup server that is fully ready to jump in production with a minimum of reconfiguration. A couple more questions, will you do much recording? and what kind of disks do you plan on using? Hope this helps, MATT--- -----Original Message----- From: Warren Smith [mailto:warren@serverplus.com] Sent: Monday, May 23, 2005 8:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inbound call center - reliability \ scalability with queues We are wanting to move off of our legacy inter-tel phoneswitch and move to VoIP and asterisk. We are looking for a new PBX because the inter-tel switch is too difficult to integrate our existing (and new) software into. We are a technical support center. All our calls currently come in on toll free numbers via T1's, and there are 3 of them. I want to use a media gateway to convert the T1's into SIP VOIP (I want reliable hardware for the gateway), and use asterisk as the PBX having all incoming and outgoing channels as SIP. Almost all dialplans will be using Queues, and there will likely be no more than 10 queues, with (currently) about 80 incoming toll-free numbers. There are approximatley 30 agents, but as of right now there are no more than 15 agents logged in at a time. We need to be able to support 60-70 simultaneous calls initially and we have to be able to do this reliably. We also need to be able to scale into the 100's of simultanous calls range. What would be the best option, to have 2 powerful machines (dual powersupply, ) with one as a hotswappable backup or have multiple machines with a sort of load balancer setup? Having multiple machines could possibly cost less, but I'm not sure how the queues and agents would be managed across multiple machines. I.e. how would the agent 'login' to each asterisk machine so that the calls could be handed to it, and how would the calls get handed to an available agent by 4 seperate asterisk machines? I've read through the wiki, but I'm not sure how much overhead queues would put into the system. I want to have all the codecs the same, so the asterisk machines doesn't do any transcoding, and have all channels as SIP. There will be music on hold. Would a dual 2.8 ghz xeon in this config be able to handle 80 simultaneous incoming calls? Would using the 711 codec make a difference in available processing power? I'm sorry if this has been answered a million times already, I just didn't see many configs close to what we're trying to do to compare to. Thanks for any input you may have. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
mattf
2005-May-24 10:41 UTC
[Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues
OK, If you are going to be recording all calls you will need to rethink things a bit. Recording calls limits you to 50-60 consecutive conversations per server before audio distortion starts to occur. You will probably want to think about limiting yourself to 3 T1s per machine. There are many ways to set this up and I think you will probably have to go through some trial-and-error before you find the perfect system layout for your operations. I would first try setting up machines that would just have the T1s on them and take the calls in(or out) and record them. Then have those connect(through IAX or T1 crossover) to the servers that have your queues and phones set up on them. You will also need some really big archiving mechanism if you want to keep those recordings around to reference in the future. audio recordings can take up a lot of space if you need to keep them for 3 years like we do. SCSI RAIDs are a great solution, but sometimes(in very loaded servers) they can hit the PCI-bus bottleneck and have issues. You may see the "ast_channel_walk_locked" warning in Asterisk when this happens. MATT--- -----Original Message----- From: Warren Smith [mailto:warren@serverplus.com] Sent: Tuesday, May 24, 2005 12:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Inbound call center - reliability \ scalability with queues I would say that we would need to be able to scale to the 200+ consecutive call range in the near future (6 months), and hopefully to the 500+ within the next two years. We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K rpm SCSI). We also are planning on recording all queue calls that are answered, and possibly outbound calls made by support agents since it has proven to be extremely helpful when it comes to training. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mattf Sent: Monday, May 23, 2005 6:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Inbound call center - reliability \ scalability with queues For an inbound call center with 4 T1s and 30-50 agents on you would do just fine with a single, one-processor machine. We have handled more than this on a single P4 server although we use astGUIclient instead of Asterisk queues, but the load is very similar. I would recommend a Sangoma Quad T1 card because they are about 30% more efficient than Digium T1 cards. When you say that you need to scale to 100s of consecutive calls, is that closer to 200 or 900? and what timeframe is that planned for? We have a distributed in/outbound call center environment across 4 geographic locations with over 20 T1s connected so it is possible for Asterisk to handle over 1000 consecutive calls across the system if you design it right. One of the reasons we don't use Asterisk queues, other than the difficulty in customizing the code to work with the ManagerAPI and client apps, is that it was hard to scale across multiple servers. That's why we use the astGUIclient suite which is more customizable and scalable across multiple servers, although (and it pains me to say this because we developed it) it is not as easy to install and setup than just creating an Asterisk queue. We use a combination of SIP, IAX and Zap client phones depending on the system and the user and yes 711 is always best to use when you can. And if you have many remote phones using a codec like GSM it may actually be better to have a dedicated machine doing nothing but the transcoding from GSM->711 and then just using IAX or a crossover Zap T1 to the inbound server to reduce processor load. In any case it is always advisable to have a backup server that is fully ready to jump in production with a minimum of reconfiguration. A couple more questions, will you do much recording? and what kind of disks do you plan on using? Hope this helps, MATT--- -----Original Message----- From: Warren Smith [mailto:warren@serverplus.com] Sent: Monday, May 23, 2005 8:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inbound call center - reliability \ scalability with queues We are wanting to move off of our legacy inter-tel phoneswitch and move to VoIP and asterisk. We are looking for a new PBX because the inter-tel switch is too difficult to integrate our existing (and new) software into. We are a technical support center. All our calls currently come in on toll free numbers via T1's, and there are 3 of them. I want to use a media gateway to convert the T1's into SIP VOIP (I want reliable hardware for the gateway), and use asterisk as the PBX having all incoming and outgoing channels as SIP. Almost all dialplans will be using Queues, and there will likely be no more than 10 queues, with (currently) about 80 incoming toll-free numbers. There are approximatley 30 agents, but as of right now there are no more than 15 agents logged in at a time. We need to be able to support 60-70 simultaneous calls initially and we have to be able to do this reliably. We also need to be able to scale into the 100's of simultanous calls range. What would be the best option, to have 2 powerful machines (dual powersupply, ) with one as a hotswappable backup or have multiple machines with a sort of load balancer setup? Having multiple machines could possibly cost less, but I'm not sure how the queues and agents would be managed across multiple machines. I.e. how would the agent 'login' to each asterisk machine so that the calls could be handed to it, and how would the calls get handed to an available agent by 4 seperate asterisk machines? I've read through the wiki, but I'm not sure how much overhead queues would put into the system. I want to have all the codecs the same, so the asterisk machines doesn't do any transcoding, and have all channels as SIP. There will be music on hold. Would a dual 2.8 ghz xeon in this config be able to handle 80 simultaneous incoming calls? Would using the 711 codec make a difference in available processing power? I'm sorry if this has been answered a million times already, I just didn't see many configs close to what we're trying to do to compare to. Thanks for any input you may have. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Ilan Rabinovitch
2005-May-24 10:45 UTC
[Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues
Matt, Are you doing any call recording / monitoring? What percentage? Ilan On 5/23/05, mattf <mattf@vicimarketing.com> wrote:> For an inbound call center with 4 T1s and 30-50 agents on you would do just > fine with a single, one-processor machine. We have handled more than this on > a single P4 server although we use astGUIclient instead of Asterisk queues, > but the load is very similar. I would recommend a Sangoma Quad T1 card > because they are about 30% more efficient than Digium T1 cards. > > When you say that you need to scale to 100s of consecutive calls, is that > closer to 200 or 900? and what timeframe is that planned for? > > We have a distributed in/outbound call center environment across 4 > geographic locations with over 20 T1s connected so it is possible for > Asterisk to handle over 1000 consecutive calls across the system if you > design it right. One of the reasons we don't use Asterisk queues, other than > the difficulty in customizing the code to work with the ManagerAPI and > client apps, is that it was hard to scale across multiple servers. That's > why we use the astGUIclient suite which is more customizable and scalable > across multiple servers, although (and it pains me to say this because we > developed it) it is not as easy to install and setup than just creating an > Asterisk queue. > > We use a combination of SIP, IAX and Zap client phones depending on the > system and the user and yes 711 is always best to use when you can. And if > you have many remote phones using a codec like GSM it may actually be better > to have a dedicated machine doing nothing but the transcoding from GSM->711 > and then just using IAX or a crossover Zap T1 to the inbound server to > reduce processor load. > > In any case it is always advisable to have a backup server that is fully > ready to jump in production with a minimum of reconfiguration. > > A couple more questions, will you do much recording? and what kind of disks > do you plan on using? > > Hope this helps, > > MATT--- > > -----Original Message----- > From: Warren Smith [mailto:warren@serverplus.com] > Sent: Monday, May 23, 2005 8:12 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Inbound call center - reliability \ scalability > with queues > > > We are wanting to move off of our legacy inter-tel phoneswitch and move to > VoIP and asterisk. We are looking for a new PBX because the inter-tel > switch is too difficult to integrate our existing (and new) software into. > > We are a technical support center. All our calls currently come in on toll > free numbers via T1's, and there are 3 of them. I want to use a media > gateway to convert the T1's into SIP VOIP (I want reliable hardware for the > gateway), and use asterisk as the PBX having all incoming and outgoing > channels as SIP. Almost all dialplans will be using Queues, and there will > likely be no more than 10 queues, with (currently) about 80 incoming > toll-free numbers. There are approximatley 30 agents, but as of right now > there are no more than 15 agents logged in at a time. We need to be able to > support 60-70 simultaneous calls initially and we have to be able to do this > reliably. We also need to be able to scale into the 100's of simultanous > calls range. > > What would be the best option, to have 2 powerful machines (dual > powersupply, ) with one as a hotswappable backup or have multiple machines > with a sort of load balancer setup? Having multiple machines could possibly > cost less, but I'm not sure how the queues and agents would be managed > across multiple machines. I.e. how would the agent 'login' to each asterisk > machine so that the calls could be handed to it, and how would the calls get > handed to an available agent by 4 seperate asterisk machines? I've read > through the wiki, but I'm not sure how much overhead queues would put into > the system. I want to have all the codecs the same, so the asterisk > machines doesn't do any transcoding, and have all channels as SIP. There > will be music on hold. Would a dual 2.8 ghz xeon in this config be able to > handle 80 simultaneous incoming calls? Would using the 711 codec make a > difference in available processing power? > > I'm sorry if this has been answered a million times already, I just didn't > see many configs close to what we're trying to do to compare to. Thanks for > any input you may have. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
mattf
2005-May-24 11:01 UTC
[Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues
We have several different setups, but on a couple servers we are doing upto 50 concurrent conversations of recording. We ran into the 50-60 recording ceiling about a year ago and it's mostly the hard drive that limits it to that number, really it's a lot if you think about it, Asterisk is having the hard drive write 100-120 audio files(-in and -out for each conversation) several times a second. It is also important to note that we mix them with sox after hours to reduce load on the system and load on the drives. Although this does mean that the recordings are not available until the next day. We also have setup 2 systems to copy the in and out files off to another machine to be mixed more closely to realtime so that is an alternative. MATT--- -----Original Message----- From: Ilan Rabinovitch [mailto:irabinovitch@gmail.com] Sent: Tuesday, May 24, 2005 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues Matt, Are you doing any call recording / monitoring? What percentage? Ilan On 5/23/05, mattf <mattf@vicimarketing.com> wrote:> For an inbound call center with 4 T1s and 30-50 agents on you would dojust> fine with a single, one-processor machine. We have handled more than thison> a single P4 server although we use astGUIclient instead of Asteriskqueues,> but the load is very similar. I would recommend a Sangoma Quad T1 card > because they are about 30% more efficient than Digium T1 cards. > > When you say that you need to scale to 100s of consecutive calls, is that > closer to 200 or 900? and what timeframe is that planned for? > > We have a distributed in/outbound call center environment across 4 > geographic locations with over 20 T1s connected so it is possible for > Asterisk to handle over 1000 consecutive calls across the system if you > design it right. One of the reasons we don't use Asterisk queues, otherthan> the difficulty in customizing the code to work with the ManagerAPI and > client apps, is that it was hard to scale across multiple servers. That's > why we use the astGUIclient suite which is more customizable and scalable > across multiple servers, although (and it pains me to say this because we > developed it) it is not as easy to install and setup than just creating an > Asterisk queue. > > We use a combination of SIP, IAX and Zap client phones depending on the > system and the user and yes 711 is always best to use when you can. And if > you have many remote phones using a codec like GSM it may actually bebetter> to have a dedicated machine doing nothing but the transcoding fromGSM->711> and then just using IAX or a crossover Zap T1 to the inbound server to > reduce processor load. > > In any case it is always advisable to have a backup server that is fully > ready to jump in production with a minimum of reconfiguration. > > A couple more questions, will you do much recording? and what kind ofdisks> do you plan on using? > > Hope this helps, > > MATT--- > > -----Original Message----- > From: Warren Smith [mailto:warren@serverplus.com] > Sent: Monday, May 23, 2005 8:12 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Inbound call center - reliability \ scalability > with queues > > > We are wanting to move off of our legacy inter-tel phoneswitch and move to > VoIP and asterisk. We are looking for a new PBX because the inter-tel > switch is too difficult to integrate our existing (and new) software into. > > We are a technical support center. All our calls currently come in ontoll> free numbers via T1's, and there are 3 of them. I want to use a media > gateway to convert the T1's into SIP VOIP (I want reliable hardware forthe> gateway), and use asterisk as the PBX having all incoming and outgoing > channels as SIP. Almost all dialplans will be using Queues, and therewill> likely be no more than 10 queues, with (currently) about 80 incoming > toll-free numbers. There are approximatley 30 agents, but as of right now > there are no more than 15 agents logged in at a time. We need to be ableto> support 60-70 simultaneous calls initially and we have to be able to dothis> reliably. We also need to be able to scale into the 100's of simultanous > calls range. > > What would be the best option, to have 2 powerful machines (dual > powersupply, ) with one as a hotswappable backup or have multiple machines > with a sort of load balancer setup? Having multiple machines couldpossibly> cost less, but I'm not sure how the queues and agents would be managed > across multiple machines. I.e. how would the agent 'login' to eachasterisk> machine so that the calls could be handed to it, and how would the callsget> handed to an available agent by 4 seperate asterisk machines? I've read > through the wiki, but I'm not sure how much overhead queues would put into > the system. I want to have all the codecs the same, so the asterisk > machines doesn't do any transcoding, and have all channels as SIP. There > will be music on hold. Would a dual 2.8 ghz xeon in this config be ableto> handle 80 simultaneous incoming calls? Would using the 711 codec make a > difference in available processing power? > > I'm sorry if this has been answered a million times already, I just didn't > see many configs close to what we're trying to do to compare to. Thanksfor> any input you may have. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
mattf
2005-May-24 13:15 UTC
[Asterisk-Users] Inbound call center - reliability \ scalabil ity with queues
Well, that really changes things then. I'm not really sure what to tell you because we've never done it that way. The ciscos are limited in how you can have them send calls to different servers based upon specific parameters so you will be limited there somewhat. Is there a specific reason you're not going to use Asterisk servers for the T1->SIP conversion? It would allow you to do the recording up front and give you more control over call handling, and I can't imagine that you can get a new 3 x T1 Cisco VOIP machine for less than $5000 like you can with Asterisk. MATT--- -----Original Message----- From: Warren Smith [mailto:warren@serverplus.com] Sent: Tuesday, May 24, 2005 2:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Inbound call center - reliability \ scalability with queues The asterisk machines will not have anything to do with the T1's, when they receive the call it will be SIP VOIP. There will be media gateways (i.e. cisco media gateways) to change all T1 signals to VOIP before it reaches the PBX. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mattf Sent: Tuesday, May 24, 2005 11:42 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Inbound call center - reliability \ scalability with queues OK, If you are going to be recording all calls you will need to rethink things a bit. Recording calls limits you to 50-60 consecutive conversations per server before audio distortion starts to occur. You will probably want to think about limiting yourself to 3 T1s per machine. There are many ways to set this up and I think you will probably have to go through some trial-and-error before you find the perfect system layout for your operations. I would first try setting up machines that would just have the T1s on them and take the calls in(or out) and record them. Then have those connect(through IAX or T1 crossover) to the servers that have your queues and phones set up on them. You will also need some really big archiving mechanism if you want to keep those recordings around to reference in the future. audio recordings can take up a lot of space if you need to keep them for 3 years like we do. SCSI RAIDs are a great solution, but sometimes(in very loaded servers) they can hit the PCI-bus bottleneck and have issues. You may see the "ast_channel_walk_locked" warning in Asterisk when this happens. MATT--- -----Original Message----- From: Warren Smith [mailto:warren@serverplus.com] Sent: Tuesday, May 24, 2005 12:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Inbound call center - reliability \ scalability with queues I would say that we would need to be able to scale to the 200+ consecutive call range in the near future (6 months), and hopefully to the 500+ within the next two years. We plan on using at least 1+0 raid with a minimum of 4 disks (minimum of 10K rpm SCSI). We also are planning on recording all queue calls that are answered, and possibly outbound calls made by support agents since it has proven to be extremely helpful when it comes to training. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mattf Sent: Monday, May 23, 2005 6:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Inbound call center - reliability \ scalability with queues For an inbound call center with 4 T1s and 30-50 agents on you would do just fine with a single, one-processor machine. We have handled more than this on a single P4 server although we use astGUIclient instead of Asterisk queues, but the load is very similar. I would recommend a Sangoma Quad T1 card because they are about 30% more efficient than Digium T1 cards. When you say that you need to scale to 100s of consecutive calls, is that closer to 200 or 900? and what timeframe is that planned for? We have a distributed in/outbound call center environment across 4 geographic locations with over 20 T1s connected so it is possible for Asterisk to handle over 1000 consecutive calls across the system if you design it right. One of the reasons we don't use Asterisk queues, other than the difficulty in customizing the code to work with the ManagerAPI and client apps, is that it was hard to scale across multiple servers. That's why we use the astGUIclient suite which is more customizable and scalable across multiple servers, although (and it pains me to say this because we developed it) it is not as easy to install and setup than just creating an Asterisk queue. We use a combination of SIP, IAX and Zap client phones depending on the system and the user and yes 711 is always best to use when you can. And if you have many remote phones using a codec like GSM it may actually be better to have a dedicated machine doing nothing but the transcoding from GSM->711 and then just using IAX or a crossover Zap T1 to the inbound server to reduce processor load. In any case it is always advisable to have a backup server that is fully ready to jump in production with a minimum of reconfiguration. A couple more questions, will you do much recording? and what kind of disks do you plan on using? Hope this helps, MATT--- -----Original Message----- From: Warren Smith [mailto:warren@serverplus.com] Sent: Monday, May 23, 2005 8:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inbound call center - reliability \ scalability with queues We are wanting to move off of our legacy inter-tel phoneswitch and move to VoIP and asterisk. We are looking for a new PBX because the inter-tel switch is too difficult to integrate our existing (and new) software into. We are a technical support center. All our calls currently come in on toll free numbers via T1's, and there are 3 of them. I want to use a media gateway to convert the T1's into SIP VOIP (I want reliable hardware for the gateway), and use asterisk as the PBX having all incoming and outgoing channels as SIP. Almost all dialplans will be using Queues, and there will likely be no more than 10 queues, with (currently) about 80 incoming toll-free numbers. There are approximatley 30 agents, but as of right now there are no more than 15 agents logged in at a time. We need to be able to support 60-70 simultaneous calls initially and we have to be able to do this reliably. We also need to be able to scale into the 100's of simultanous calls range. What would be the best option, to have 2 powerful machines (dual powersupply, ) with one as a hotswappable backup or have multiple machines with a sort of load balancer setup? Having multiple machines could possibly cost less, but I'm not sure how the queues and agents would be managed across multiple machines. I.e. how would the agent 'login' to each asterisk machine so that the calls could be handed to it, and how would the calls get handed to an available agent by 4 seperate asterisk machines? I've read through the wiki, but I'm not sure how much overhead queues would put into the system. I want to have all the codecs the same, so the asterisk machines doesn't do any transcoding, and have all channels as SIP. There will be music on hold. Would a dual 2.8 ghz xeon in this config be able to handle 80 simultaneous incoming calls? Would using the 711 codec make a difference in available processing power? I'm sorry if this has been answered a million times already, I just didn't see many configs close to what we're trying to do to compare to. Thanks for any input you may have. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users