Re: [Asterisk-Users] Asterisk Hardware Recommendation
Daniel Salama wrote: This is great information. I have the following questions based on a hypothetical scenario and some assumptions: Based on the price of these configurations, I wouldn't even mind putting two servers each with 2 T1s just so that I could get all calls recorded and distribute the risk of failure. Now, I don't know if it would make a difference or not, but here it goes: Assuming the cost of the systems is of no importance for a moment (actually looking for the most scalable and reliable solution), which would be a better approach to solve the issue of activating 4 T1s which will be constantly taxed with load and be able to record all conversations: Scenario 1: 4 T1s into Asterisk (A1) where all SIP agents register. Call recording in A1. PSTN --4xT1-- A1 SIP_Agents Scenario 2: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where all SIP agents register (IAX to SIP transcoding). Call recording in A1 or A2. PSTN --4xT1-- A1 A2 SIP_Agents Scenario 3: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where half of SIP agents register to, and the other half would register in A1. Call recording in A1 and/or A2. PSTN --4xT1-- A1 SIP_Agents A1 --IAX-- A2 SIP_Agents Scenario 4: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisk (A2) will connect to A1 and A3 via IAX. All SIP Agents register at A2 (IAX to SIP transcoding). Call recording in [A1 and A3] or A2. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A2 SIP_Agents Scenario 5: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisks (A2 and A4) will connect to A1 and A3 respectively via IAX. Half SIP Agents register in A2 and other half in A4 (IAX to SIP transcoding). Call recording in [A1 and A3] or [A2 and A4]. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A4 SIP_Agents Hopefully you're all able to understand my 5 scenarios. I guess, my questions would be: 1) Is there a load limiting factor in terms of whether you do the Monitoring of the calls when you're doing TDM-IAX transcoding or IAX-SIP transcoding? 2) Will a single CPU machine handle the 4 T1s to do TDM-IAX transcoding, if another machine is doing the actual recording (IAX-SIP transconding) (Scenarios 2,3,4,5). Basically, just setup cheap Asterisk boxes to act as VoIP gateways and the distribute the load and/or intelligence on other Asterisk boxes to connect SIP agents and all dialing rules, etc? I haven't seen this before--can an agent log into a queue on a remote (i.e. over IAX) Asterisk server? ___ 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
RE: [Asterisk-Users] Asterisk Hardware Recommendation
If price would truly not an option just get one of the Signate Telephony 5000 servers(http://www.signate.com/pbx.php) They are about $18,000 and allow you to have upto 5000 SIP streams go through it. You could have that be your gateway and do the SIP-IAX through that machine and scale upto 100 T1s if you want. But that is a bit steep. So on to your choices. I would really say that the setup you choose will depend on what kind of users you have as well as how often you need to change/add users to the system and how the users are using the system at what times. Any of them that you listed could work depending on how they are used, but in some cases you may not want to use some of the scenarios listed because they would either be incapable of meeting your needs or overly complex to manage. The easiest and cheapest one would actually not be listed: Scenario 6: Direct SIP-Zap on two separate servers half SIP users on each server PSTN --2xT1-- A1 SIP_Agents PSTN --2xT1-- A2 SIP_Agents There is really no reason to have another 2 servers running IAX to the T1 servers, and this is simple and easy to set up and involves only 2 machines. The next setup I would recommend would be Scenario 4, although you will have to get a machine with a fast/wide BUS(like an Apple G5) to handle ever increasing numbers of SIP-IAX streams as the system would grow. If you can explain more about what kind of use this system will have I can give a better recommendation. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation This is great information. I have the following questions based on a hypothetical scenario and some assumptions: Based on the price of these configurations, I wouldn't even mind putting two servers each with 2 T1s just so that I could get all calls recorded and distribute the risk of failure. Now, I don't know if it would make a difference or not, but here it goes: Assuming the cost of the systems is of no importance for a moment (actually looking for the most scalable and reliable solution), which would be a better approach to solve the issue of activating 4 T1s which will be constantly taxed with load and be able to record all conversations: Scenario 1: 4 T1s into Asterisk (A1) where all SIP agents register. Call recording in A1. PSTN --4xT1-- A1 SIP_Agents Scenario 2: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where all SIP agents register (IAX to SIP transcoding). Call recording in A1 or A2. PSTN --4xT1-- A1 A2 SIP_Agents Scenario 3: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where half of SIP agents register to, and the other half would register in A1. Call recording in A1 and/or A2. PSTN --4xT1-- A1 SIP_Agents A1 --IAX-- A2 SIP_Agents Scenario 4: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisk (A2) will connect to A1 and A3 via IAX. All SIP Agents register at A2 (IAX to SIP transcoding). Call recording in [A1 and A3] or A2. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A2 SIP_Agents Scenario 5: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisks (A2 and A4) will connect to A1 and A3 respectively via IAX. Half SIP Agents register in A2 and other half in A4 (IAX to SIP transcoding). Call recording in [A1 and A3] or [A2 and A4]. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A4 SIP_Agents Hopefully you're all able to understand my 5 scenarios. I guess, my questions would be: 1) Is there a load limiting factor in terms of whether you do the Monitoring of the calls when you're doing TDM-IAX transcoding or IAX-SIP transcoding? 2) Will a single CPU machine handle the 4 T1s to do TDM-IAX transcoding, if another machine is doing the actual recording (IAX-SIP transconding) (Scenarios 2,3,4,5). Basically, just setup cheap Asterisk boxes to act as VoIP gateways and the distribute the load and/or intelligence on other Asterisk boxes to connect SIP agents and all dialing rules, etc? Thanks, Daniel On Apr 28, 2005, at 9:17 PM, mattf wrote: You can throw together a single P4 3GHz with 1GB RAM and 2 x 80GB SATA HD for about $600. One of those can easily handle a Sangoma dual T1 card($900) or a Digium quad T1 card($1400). For that you can have a system for about $1500-$2000 that will be able to fully record 2 T1s(48 channels) worth of Zap-SIP conversations. Putting two of those together with a nice big fileserver will give you a lot of flexibility, as well as only a reduction in capacity if one of the servers go down instead of a total outage, for about the same overall price of a single high-end Dual Xeon server. Building your system
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Sure. I setup a small lab on a machine with 4 T1s and 36 agents logged in. The system was configured to Monitor all outbound calls as well as monitor all calls distributed by Queue app (monitor-format setting in queues.conf). When recording to local disk, everything was working fine. Agents were busy 99.5% and there were at least 30 calls waiting in Queue to be distributed. Average call conversation length was about 7.5 minutes. Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E. The moment we pushed the load on the Asterisk machine, everything worked for about 40 seconds. Then call quality started suffering significantly. Chopped audio. Bad audio. No audio. Good audio. You could imagine. So we stopped the test. Then we unmounted the NFS drive and repeated the test again. Everything worked fine again. The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. During all tests, CPU utilization was about 55% on the average (for each CPU). Memory usage was under 1G. I would say I need to try more troubleshooting. Maybe there was congestion on the Fast-E, although preliminary analysis indicates there were no CRC errors, collisions, or packet loss. The NFS machine was completely idle. Last, we repeated the test over a 1 hour period. This time, Monitor was recording on local drives and we were copying files every 15 minutes with a background process (perl script) to NFS mount point. Everything worked fine as well. I don't know if these tests are conclusive yet. However, from the results so far, I would recommend staying away from recording to NFS mounted point. I will continue running simulations to see if anything else can be identified. Thanks, Daniel On Apr 28, 2005, at 7:26 PM, Matt Roth wrote: Daniel, Could you expand upon your experience recording to an NFS mounted drive. We are looking to use a TDM-VoIP gateway to route 16+ spans to a single Asterisk server. We were hoping to Monitor using the following scheme: - Monitor application executed on Asterisk server (no 'm' flag) - Pick a codec that the Monitor application can record in natively so that no transcoding is done on the leg files (can this be done?) - Record the leg files to an NFS mounted drive on a remote machine - Have soxmix take care of mixing and transcoding the leg files into the desired format on the remote machine According to you this now looks like a VERY BAD IDEA. Does anyone out there have any experience using monitor to digitally record large numbers of spans (16+) on a single Asterisk server using a VoIP gateway instead of TDM cards? Is it feasible? We are trying to keep the Asterisk server as slim as possible, but would like to stick to one box so that we can have centralized queuing, configuration, and reporting. Has anyone had any luck using Monitor to record to an NFS mounted drive? Are there any other options to remove the overhead of the disk subsystem when recording calls? Thanks, Matthew Roth http://voip-info.org/tiki-index.php? page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Thanks, Daniel On Apr 28, 2005, at 6:42 PM, mattf wrote: I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA drives, with a nice big archive server that the audio will be copied to several times a day. Also, do not record(Monitor) with the 'm' flag on because this will also lead to more disk read-write while you are already trying to write another 100 or so streams. Offload the -in and -out files to the archive server and let it soxmix them together instead. This is the method that we have settled on for our 12 Asterisk servers and it works rather well for us. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Hardware Recommendation Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Well, I don't think I'm ready to spend that much money :) I understand your point regarding that load depends on usage. SIP_Agents are simply agents answering calls. Average call length would be about 8 minutes. During some of these calls (maybe 25%), agents will conference the call (PSTN channel) with internal IVR script. I like Scenario 6. Will look into that further. However, if the above information gives you more grounds to make additional comments, please do so :) Thanks, Daniel On Apr 29, 2005, at 10:21 AM, mattf wrote: If price would truly not an option just get one of the Signate Telephony 5000 servers(http://www.signate.com/pbx.php) They are about $18,000 and allow you to have upto 5000 SIP streams go through it. You could have that be your gateway and do the SIP-IAX through that machine and scale upto 100 T1s if you want. But that is a bit steep. So on to your choices. I would really say that the setup you choose will depend on what kind of users you have as well as how often you need to change/add users to the system and how the users are using the system at what times. Any of them that you listed could work depending on how they are used, but in some cases you may not want to use some of the scenarios listed because they would either be incapable of meeting your needs or overly complex to manage. The easiest and cheapest one would actually not be listed: Scenario 6: Direct SIP-Zap on two separate servers half SIP users on each server PSTN --2xT1-- A1 SIP_Agents PSTN --2xT1-- A2 SIP_Agents There is really no reason to have another 2 servers running IAX to the T1 servers, and this is simple and easy to set up and involves only 2 machines. The next setup I would recommend would be Scenario 4, although you will have to get a machine with a fast/wide BUS(like an Apple G5) to handle ever increasing numbers of SIP-IAX streams as the system would grow. If you can explain more about what kind of use this system will have I can give a better recommendation. MATT--- ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Daniel, Thanks alot for this post. You were right on time with valuable information. Thanks again, Steve - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 12:37 PM Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation Sure. I setup a small lab on a machine with 4 T1s and 36 agents logged in. The system was configured to Monitor all outbound calls as well as monitor all calls distributed by Queue app (monitor-format setting in queues.conf). When recording to local disk, everything was working fine. Agents were busy 99.5% and there were at least 30 calls waiting in Queue to be distributed. Average call conversation length was about 7.5 minutes. Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E. The moment we pushed the load on the Asterisk machine, everything worked for about 40 seconds. Then call quality started suffering significantly. Chopped audio. Bad audio. No audio. Good audio. You could imagine. So we stopped the test. Then we unmounted the NFS drive and repeated the test again. Everything worked fine again. The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. During all tests, CPU utilization was about 55% on the average (for each CPU). Memory usage was under 1G. I would say I need to try more troubleshooting. Maybe there was congestion on the Fast-E, although preliminary analysis indicates there were no CRC errors, collisions, or packet loss. The NFS machine was completely idle. Last, we repeated the test over a 1 hour period. This time, Monitor was recording on local drives and we were copying files every 15 minutes with a background process (perl script) to NFS mount point. Everything worked fine as well. I don't know if these tests are conclusive yet. However, from the results so far, I would recommend staying away from recording to NFS mounted point. I will continue running simulations to see if anything else can be identified. Thanks, Daniel On Apr 28, 2005, at 7:26 PM, Matt Roth wrote: Daniel, Could you expand upon your experience recording to an NFS mounted drive. We are looking to use a TDM-VoIP gateway to route 16+ spans to a single Asterisk server. We were hoping to Monitor using the following scheme: - Monitor application executed on Asterisk server (no 'm' flag) - Pick a codec that the Monitor application can record in natively so that no transcoding is done on the leg files (can this be done?) - Record the leg files to an NFS mounted drive on a remote machine - Have soxmix take care of mixing and transcoding the leg files into the desired format on the remote machine According to you this now looks like a VERY BAD IDEA. Does anyone out there have any experience using monitor to digitally record large numbers of spans (16+) on a single Asterisk server using a VoIP gateway instead of TDM cards? Is it feasible? We are trying to keep the Asterisk server as slim as possible, but would like to stick to one box so that we can have centralized queuing, configuration, and reporting. Has anyone had any luck using Monitor to record to an NFS mounted drive? Are there any other options to remove the overhead of the disk subsystem when recording calls? Thanks, Matthew Roth http://voip-info.org/tiki-index.php? page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Thanks, Daniel On Apr 28, 2005, at 6:42 PM, mattf wrote: I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA drives, with a nice big archive server that the audio will be copied to several times a day. Also, do not record(Monitor) with the 'm' flag on because this will also lead to more disk read-write while you are already trying to write another 100 or so streams. Offload the -in and -out files to the archive server and let it soxmix them together instead. This is the method that we have settled on for our 12 Asterisk servers and it works rather well for us. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 5:56 PM
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Thanks Daniel, We may end up replicating your tests in order to confirm some of your results. I don't know if it will be anytime soon, because we don't have the hardware yet. Regardless, I will share my results with the list. Anyone out there have any ideas on why the NFS mount affected call quality? It seems backwards, since it should have relieved some of the load from the Asterisk machine. Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Sure. I setup a small lab on a machine with 4 T1s and 36 agents logged in. The system was configured to Monitor all outbound calls as well as monitor all calls distributed by Queue app (monitor-format setting in queues.conf). When recording to local disk, everything was working fine. Agents were busy 99.5% and there were at least 30 calls waiting in Queue to be distributed. Average call conversation length was about 7.5 minutes. Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E. The moment we pushed the load on the Asterisk machine, everything worked for about 40 seconds. Then call quality started suffering significantly. Chopped audio. Bad audio. No audio. Good audio. You could imagine. So we stopped the test. Then we unmounted the NFS drive and repeated the test again. Everything worked fine again. The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. During all tests, CPU utilization was about 55% on the average (for each CPU). Memory usage was under 1G. I would say I need to try more troubleshooting. Maybe there was congestion on the Fast-E, although preliminary analysis indicates there were no CRC errors, collisions, or packet loss. The NFS machine was completely idle. Last, we repeated the test over a 1 hour period. This time, Monitor was recording on local drives and we were copying files every 15 minutes with a background process (perl script) to NFS mount point. Everything worked fine as well. I don't know if these tests are conclusive yet. However, from the results so far, I would recommend staying away from recording to NFS mounted point. I will continue running simulations to see if anything else can be identified. Thanks, Daniel ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Does anyone have experience with using NAS (http://en.wikipedia.org/wiki/Network-attached_storage) or SAN (http://en.wikipedia.org/wiki/Storage_area_network) for this application? Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Sure. I setup a small lab on a machine with 4 T1s and 36 agents logged in. The system was configured to Monitor all outbound calls as well as monitor all calls distributed by Queue app (monitor-format setting in queues.conf). When recording to local disk, everything was working fine. Agents were busy 99.5% and there were at least 30 calls waiting in Queue to be distributed. Average call conversation length was about 7.5 minutes. Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E. The moment we pushed the load on the Asterisk machine, everything worked for about 40 seconds. Then call quality started suffering significantly. Chopped audio. Bad audio. No audio. Good audio. You could imagine. So we stopped the test. Then we unmounted the NFS drive and repeated the test again. Everything worked fine again. The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. During all tests, CPU utilization was about 55% on the average (for each CPU). Memory usage was under 1G. I would say I need to try more troubleshooting. Maybe there was congestion on the Fast-E, although preliminary analysis indicates there were no CRC errors, collisions, or packet loss. The NFS machine was completely idle. Last, we repeated the test over a 1 hour period. This time, Monitor was recording on local drives and we were copying files every 15 minutes with a background process (perl script) to NFS mount point. Everything worked fine as well. I don't know if these tests are conclusive yet. However, from the results so far, I would recommend staying away from recording to NFS mounted point. I will continue running simulations to see if anything else can be identified. Thanks, Daniel ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Maybe something like this would be good. http://www.pcmicrostore.com/PartDetail.aspx?q=p:10502197 - Original Message - From: Matt Roth [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 2:11 PM Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation Does anyone have experience with using NAS (http://en.wikipedia.org/wiki/Network-attached_storage) or SAN (http://en.wikipedia.org/wiki/Storage_area_network) for this application? Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Sure. I setup a small lab on a machine with 4 T1s and 36 agents logged in. The system was configured to Monitor all outbound calls as well as monitor all calls distributed by Queue app (monitor-format setting in queues.conf). When recording to local disk, everything was working fine. Agents were busy 99.5% and there were at least 30 calls waiting in Queue to be distributed. Average call conversation length was about 7.5 minutes. Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E. The moment we pushed the load on the Asterisk machine, everything worked for about 40 seconds. Then call quality started suffering significantly. Chopped audio. Bad audio. No audio. Good audio. You could imagine. So we stopped the test. Then we unmounted the NFS drive and repeated the test again. Everything worked fine again. The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. During all tests, CPU utilization was about 55% on the average (for each CPU). Memory usage was under 1G. I would say I need to try more troubleshooting. Maybe there was congestion on the Fast-E, although preliminary analysis indicates there were no CRC errors, collisions, or packet loss. The NFS machine was completely idle. Last, we repeated the test over a 1 hour period. This time, Monitor was recording on local drives and we were copying files every 15 minutes with a background process (perl script) to NFS mount point. Everything worked fine as well. I don't know if these tests are conclusive yet. However, from the results so far, I would recommend staying away from recording to NFS mounted point. I will continue running simulations to see if anything else can be identified. Thanks, Daniel ___ 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
RE: [Asterisk-Users] Asterisk Hardware Recommendation
Hi Matt, Does anyone have experience with using NAS (http://en.wikipedia.org/wiki/Network-attached_storage) or SAN (http://en.wikipedia.org/wiki/Storage_area_network) for this application? I've had our agent/queue recordings dumped both to local disk and SAN (currently using local disk as the SAN is being used for some other stuff). With both SAN (2GB FC) and local disk, we haven't had any problems like the ones described by Daniel. One of our live servers has 4 PRI's going with an average of about 40-50 calls at any given time during the day (60-70 peak), all being recorded, and we've had zero issues. The other two servers have similar configurations, but lower call volumes (5-20 calls depending on time of day). I'd be leary about doing it over NFS or Samba or any other sort of networked filesystem though. For our servers, that'd be extra I/O that'd have to go over either one of the network interfaces (both of which are plenty busy already with IAX2 and/or SIP). I guess it depends on your network card and how well behaved it is in terms of interrupts/etc.. You could say the same thing for local disk if you had slower drives and/or disk controllers. Ken. ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
This is an interesting question. I haven't tested it but would love to know if it works or not. Anyone? - Daniel On Apr 29, 2005, at 3:38 AM, Michael Welter wrote: I haven't seen this before--can an agent log into a queue on a remote (i.e. over IAX) Asterisk server? ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
I just read a great paper that said turn off anything that won't be used. Serial, USB , Printer ports, ETC. No Xwindows! Daniel Salama wrote: Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full capacity (96 simultaneous calls). This box will also deliver these calls to SIP users and record all their conversations via Monitor. I've heard that it's not necessarily a matter of memory (RAM) nor the need to have a multi-processor machine. But what really matters is that the motherboard (architecture) is designed to handle such a high amount of interrupts generated by the TE4XXP, the NIC, the storage array (whether it's SCSI or IDE or SATA). Does anyone have experience with particular brands of either motherboards they recommend are capable to handle this or complete systems (e.g. Dell or whichever brands), that are ready for this? Thanks, Daniel ___ 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
RE: [Asterisk-Users] Asterisk Hardware Recommendation
I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA drives, with a nice big archive server that the audio will be copied to several times a day. Also, do not record(Monitor) with the 'm' flag on because this will also lead to more disk read-write while you are already trying to write another 100 or so streams. Offload the -in and -out files to the archive server and let it soxmix them together instead. This is the method that we have settled on for our 12 Asterisk servers and it works rather well for us. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Hardware Recommendation Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full capacity (96 simultaneous calls). This box will also deliver these calls to SIP users and record all their conversations via Monitor. I've heard that it's not necessarily a matter of memory (RAM) nor the need to have a multi-processor machine. But what really matters is that the motherboard (architecture) is designed to handle such a high amount of interrupts generated by the TE4XXP, the NIC, the storage array (whether it's SCSI or IDE or SATA). Does anyone have experience with particular brands of either motherboards they recommend are capable to handle this or complete systems (e.g. Dell or whichever brands), that are ready for this? Thanks, Daniel ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Could you point me in the direction where you read that? Maybe there is more there to read. Thanks, Daniel On Apr 28, 2005, at 6:31 PM, Michael D Schelin wrote: I just read a great paper that said turn off anything that won't be used. Serial, USB , Printer ports, ETC. No Xwindows! Daniel Salama wrote: Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full capacity (96 simultaneous calls). This box will also deliver these calls to SIP users and record all their conversations via Monitor. I've heard that it's not necessarily a matter of memory (RAM) nor the need to have a multi-processor machine. But what really matters is that the motherboard (architecture) is designed to handle such a high amount of interrupts generated by the TE4XXP, the NIC, the storage array (whether it's SCSI or IDE or SATA). Does anyone have experience with particular brands of either motherboards they recommend are capable to handle this or complete systems (e.g. Dell or whichever brands), that are ready for this? Thanks, Daniel ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Thanks, Daniel On Apr 28, 2005, at 6:42 PM, mattf wrote: I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA drives, with a nice big archive server that the audio will be copied to several times a day. Also, do not record(Monitor) with the 'm' flag on because this will also lead to more disk read-write while you are already trying to write another 100 or so streams. Offload the -in and -out files to the archive server and let it soxmix them together instead. This is the method that we have settled on for our 12 Asterisk servers and it works rather well for us. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Hardware Recommendation Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full capacity (96 simultaneous calls). This box will also deliver these calls to SIP users and record all their conversations via Monitor. I've heard that it's not necessarily a matter of memory (RAM) nor the need to have a multi-processor machine. But what really matters is that the motherboard (architecture) is designed to handle such a high amount of interrupts generated by the TE4XXP, the NIC, the storage array (whether it's SCSI or IDE or SATA). Does anyone have experience with particular brands of either motherboards they recommend are capable to handle this or complete systems (e.g. Dell or whichever brands), that are ready for this? Thanks, Daniel ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Daniel Salama wrote: Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Can you share with us what happened with the NFS setup. I would have guessed this was viable provided you had ample bandwidth on your LAN. -- Andres Network Admin http://www.telesip.net ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Daniel, Could you expand upon your experience recording to an NFS mounted drive. We are looking to use a TDM-VoIP gateway to route 16+ spans to a single Asterisk server. We were hoping to Monitor using the following scheme: - Monitor application executed on Asterisk server (no 'm' flag) - Pick a codec that the Monitor application can record in natively so that no transcoding is done on the leg files (can this be done?) - Record the leg files to an NFS mounted drive on a remote machine - Have soxmix take care of mixing and transcoding the leg files into the desired format on the remote machine According to you this now looks like a VERY BAD IDEA. Does anyone out there have any experience using monitor to digitally record large numbers of spans (16+) on a single Asterisk server using a VoIP gateway instead of TDM cards? Is it feasible? We are trying to keep the Asterisk server as slim as possible, but would like to stick to one box so that we can have centralized queuing, configuration, and reporting. Has anyone had any luck using Monitor to record to an NFS mounted drive? Are there any other options to remove the overhead of the disk subsystem when recording calls? Thanks, Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Thanks, Daniel On Apr 28, 2005, at 6:42 PM, mattf wrote: I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA drives, with a nice big archive server that the audio will be copied to several times a day. Also, do not record(Monitor) with the 'm' flag on because this will also lead to more disk read-write while you are already trying to write another 100 or so streams. Offload the -in and -out files to the archive server and let it soxmix them together instead. This is the method that we have settled on for our 12 Asterisk servers and it works rather well for us. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Hardware Recommendation Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full capacity (96 simultaneous calls). This box will also deliver these calls to SIP users and record all their conversations via Monitor. I've heard that it's not necessarily a matter of memory (RAM) nor the need to have a multi-processor machine. But what really matters is that the motherboard (architecture) is designed to handle such a high amount of interrupts generated by the TE4XXP, the NIC, the storage array (whether it's SCSI or IDE or SATA). Does anyone have experience with particular brands of either motherboards they recommend are capable to handle this or complete systems (e.g. Dell or whichever brands), that are ready for this? Thanks, Daniel ___ 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
RE: [Asterisk-Users] Asterisk Hardware Recommendation
You can throw together a single P4 3GHz with 1GB RAM and 2 x 80GB SATA HD for about $600. One of those can easily handle a Sangoma dual T1 card($900) or a Digium quad T1 card($1400). For that you can have a system for about $1500-$2000 that will be able to fully record 2 T1s(48 channels) worth of Zap-SIP conversations. Putting two of those together with a nice big fileserver will give you a lot of flexibility, as well as only a reduction in capacity if one of the servers go down instead of a total outage, for about the same overall price of a single high-end Dual Xeon server. Building your system this way from the start will also allow it to scale much more easily than using just a single very expensive server. You can just add another 2 T1s of capacity at any time for just $1500. I recommend only 50 or less recordings concurrently because that is the ceiling that we discovered while trying Zap-SIP recording on both Dual Processor server-class systems and single processor cheaper commodity computers as well as on SCSI, IDE and SATA drives. If anyone out the has reliabily done recording of more than 50 conversations I would like to know the hardware architecture of your setup. Thanks, MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 6:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Thanks, Daniel On Apr 28, 2005, at 6:42 PM, mattf wrote: I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA drives, with a nice big archive server that the audio will be copied to several times a day. Also, do not record(Monitor) with the 'm' flag on because this will also lead to more disk read-write while you are already trying to write another 100 or so streams. Offload the -in and -out files to the archive server and let it soxmix them together instead. This is the method that we have settled on for our 12 Asterisk servers and it works rather well for us. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Hardware Recommendation Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full capacity (96 simultaneous calls). This box will also deliver these calls to SIP users and record all their conversations via Monitor. I've heard that it's not necessarily a matter of memory (RAM) nor the need to have a multi-processor machine. But what really matters is that the motherboard (architecture) is designed to handle such a high amount of interrupts generated by the TE4XXP, the NIC, the storage array (whether it's SCSI or IDE or SATA). Does anyone have experience with particular brands of either motherboards they recommend are capable to handle this or complete systems (e.g. Dell or whichever brands), that are ready for this? Thanks, Daniel ___ 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
Re: [Asterisk-Users] Asterisk Hardware Recommendation
This is great information. I have the following questions based on a hypothetical scenario and some assumptions: Based on the price of these configurations, I wouldn't even mind putting two servers each with 2 T1s just so that I could get all calls recorded and distribute the risk of failure. Now, I don't know if it would make a difference or not, but here it goes: Assuming the cost of the systems is of no importance for a moment (actually looking for the most scalable and reliable solution), which would be a better approach to solve the issue of activating 4 T1s which will be constantly taxed with load and be able to record all conversations: Scenario 1: 4 T1s into Asterisk (A1) where all SIP agents register. Call recording in A1. PSTN --4xT1-- A1 SIP_Agents Scenario 2: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where all SIP agents register (IAX to SIP transcoding). Call recording in A1 or A2. PSTN --4xT1-- A1 A2 SIP_Agents Scenario 3: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where half of SIP agents register to, and the other half would register in A1. Call recording in A1 and/or A2. PSTN --4xT1-- A1 SIP_Agents A1 --IAX-- A2 SIP_Agents Scenario 4: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisk (A2) will connect to A1 and A3 via IAX. All SIP Agents register at A2 (IAX to SIP transcoding). Call recording in [A1 and A3] or A2. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A2 SIP_Agents Scenario 5: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisks (A2 and A4) will connect to A1 and A3 respectively via IAX. Half SIP Agents register in A2 and other half in A4 (IAX to SIP transcoding). Call recording in [A1 and A3] or [A2 and A4]. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A4 SIP_Agents Hopefully you're all able to understand my 5 scenarios. I guess, my questions would be: 1) Is there a load limiting factor in terms of whether you do the Monitoring of the calls when you're doing TDM-IAX transcoding or IAX-SIP transcoding? 2) Will a single CPU machine handle the 4 T1s to do TDM-IAX transcoding, if another machine is doing the actual recording (IAX-SIP transconding) (Scenarios 2,3,4,5). Basically, just setup cheap Asterisk boxes to act as VoIP gateways and the distribute the load and/or intelligence on other Asterisk boxes to connect SIP agents and all dialing rules, etc? Thanks, Daniel On Apr 28, 2005, at 9:17 PM, mattf wrote: You can throw together a single P4 3GHz with 1GB RAM and 2 x 80GB SATA HD for about $600. One of those can easily handle a Sangoma dual T1 card($900) or a Digium quad T1 card($1400). For that you can have a system for about $1500-$2000 that will be able to fully record 2 T1s(48 channels) worth of Zap-SIP conversations. Putting two of those together with a nice big fileserver will give you a lot of flexibility, as well as only a reduction in capacity if one of the servers go down instead of a total outage, for about the same overall price of a single high-end Dual Xeon server. Building your system this way from the start will also allow it to scale much more easily than using just a single very expensive server. You can just add another 2 T1s of capacity at any time for just $1500. I recommend only 50 or less recordings concurrently because that is the ceiling that we discovered while trying Zap-SIP recording on both Dual Processor server-class systems and single processor cheaper commodity computers as well as on SCSI, IDE and SATA drives. If anyone out the has reliabily done recording of more than 50 conversations I would like to know the hardware architecture of your setup. Thanks, MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 6:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Thanks, Daniel On Apr 28, 2005, at 6:42 PM, mattf wrote: I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA
