Re: [asterisk-users] I need a second opinion on a new phone system deployment
Thanks again to everyone that's responded thus far. I have once again bundled the questions and answers into a single email, and am responding below. On 6/14/2013 9:43 AM, Nunya Biznatch wrote: Howdy All, They say opinions are like belly buttons, everybody has one. (that's the clean version of the saying). So I'm asking for yours. I hope you see it as a fun exercise. I'm designing a phone system from the ground up. Will be about 1000-1300 seats mixed 80/20 VoIP/Analog. 58-acre campus environment with 23 buildings. Userbase is emergency services organization, 24/7/365 operation. Down time is not an option, but blips are acceptable. Repair time is immediate. We need failover for the failover essentially. However, money is a major factor, so I have to do it all for nothing. So here's what I'm thinking. Please throw in your 2 cents. Network will be separate for phones. Fiber infrastructure available between buildings as well as copper. Internet access will be limited to a single administrative console on a temporary basis, and then only when remote 3rd party support is required. Access for 3rd party support will be supervised through remote access tools such as VNC, GoToMeeting, etc... etc... System will have zero access to local data network. This means all ancillary support servers such as DHCP, DNS, NTP, FTP, etc...etc... will be specific to the phone system. Yes, I know some responders at this time will become fixated on me gaining this connectivity. It ain't gonna happen. It's not an option. Period, end of story. These are the parameters I must work within. Trying to fix that will be a non-starter. The phone system will upgrade an existing TDM-based system. Mitel SX2000 with NuPoint Voicemail. This will not be a dump-trunk replacement. I expect at least a one to two-year transition, meaning we will have time to find problems, work bugs, and learn over time, with minimized impacts. It also means we'll be supporting two systems for some time. PBX is 97% serving your basic phone on the desk. Nothing special. Customers expect the usual list of features. There will be a goodly number of hints required for BLF on maybe 150 phones. There is one office of about 30 phones in a call-center environment that will need that service. They would be considered low volume (but don't tell them that). My Skills... I am not a Linux kung fu master, but I have built and managed my share of Linux servers on mutiple Linux flavors. I am a DCAA, having been through formal training, and have been playing with Asterisk for years, but always in fits and spurts and never in a live environment so I am by no means a kung fu master there either. I have started dabbling with virtualizations via XEN, but I am not comfortable enough with it to go live this first round. I can see myself implementing it in about three years once we're totally comfortable with what we have, so I can then have time to get that skill sorted. I was a network engineer for the US no3. telecom for a number of years, 10-years in comm-electronics in the military before that. Telecom my entire career. I've got the kung-fu to handle the network side of the house, and having administrated multiple PBXs for decade-plus, I've got the concepts down. No plans to build databases for things like directories, etc... I'm not greatly confident in those skills, and to date, haven't found anything that really stands out that would make me require that. You may think otherwise, so please chime in. I say that, but at the same time I recognize I may require a GUI interface once fully deployed to allow lower-skilled people to follow the motions to complete simple moves, adds, and changes. I'm fighting the uphill battle that is the GUI is new, CLI is old mentality. System will use G.722 for VoIP Phones. So there's the groundwork. Here's the hardware plan. Plan is to build my own servers following industry standards (ATX) and using industry standard equipment. Why? Spares? Whether redundant or not, I will still have spares for the most common elements on the shelf so equipment can be returned to service as quickly as possible. This will also allow me to be comfortable with more basic server configurations and help keep cost down. For example, Servers with single power supplies vs. dual. Also, components will be standardized for all equipment to aid in supply requirements. First the layout. 2-servers acting as gateways. Each handling 2 PRIs for outside trunks. They'll also handle the analog ports. Failover will be in the form of degraded trunk access if one should fail, but the second will be able to support services in degraded fashion. 2-servers acting as VoIP PBX. A primary and a spare. Meaning one will be capable of handling the load of the entire system, and the other will pickup when the other dies, an active/passive cluster. Will also take care of voicemail. Use of heartbeat, pacemaker
Re: [asterisk-users] I need a second opinion on a new phone system deployment
Thanks to everyone for the responses. I really appreciate it. I'll answer all questions and suggestions in this one email. (at the bottom) On 6/14/2013 9:43 AM, Nunya Biznatch wrote: Howdy All, They say opinions are like belly buttons, everybody has one. (that's the clean version of the saying). So I'm asking for yours. I hope you see it as a fun exercise. I'm designing a phone system from the ground up. Will be about 1000-1300 seats mixed 80/20 VoIP/Analog. 58-acre campus environment with 23 buildings. Userbase is emergency services organization, 24/7/365 operation. Down time is not an option, but blips are acceptable. Repair time is immediate. We need failover for the failover essentially. However, money is a major factor, so I have to do it all for nothing. So here's what I'm thinking. Please throw in your 2 cents. Network will be separate for phones. Fiber infrastructure available between buildings as well as copper. Internet access will be limited to a single administrative console on a temporary basis, and then only when remote 3rd party support is required. Access for 3rd party support will be supervised through remote access tools such as VNC, GoToMeeting, etc... etc... System will have zero access to local data network. This means all ancillary support servers such as DHCP, DNS, NTP, FTP, etc...etc... will be specific to the phone system. Yes, I know some responders at this time will become fixated on me gaining this connectivity. It ain't gonna happen. It's not an option. Period, end of story. These are the parameters I must work within. Trying to fix that will be a non-starter. The phone system will upgrade an existing TDM-based system. Mitel SX2000 with NuPoint Voicemail. This will not be a dump-trunk replacement. I expect at least a one to two-year transition, meaning we will have time to find problems, work bugs, and learn over time, with minimized impacts. It also means we'll be supporting two systems for some time. PBX is 97% serving your basic phone on the desk. Nothing special. Customers expect the usual list of features. There will be a goodly number of hints required for BLF on maybe 150 phones. There is one office of about 30 phones in a call-center environment that will need that service. They would be considered low volume (but don't tell them that). My Skills... I am not a Linux kung fu master, but I have built and managed my share of Linux servers on mutiple Linux flavors. I am a DCAA, having been through formal training, and have been playing with Asterisk for years, but always in fits and spurts and never in a live environment so I am by no means a kung fu master there either. I have started dabbling with virtualizations via XEN, but I am not comfortable enough with it to go live this first round. I can see myself implementing it in about three years once we're totally comfortable with what we have, so I can then have time to get that skill sorted. I was a network engineer for the US no3. telecom for a number of years, 10-years in comm-electronics in the military before that. Telecom my entire career. I've got the kung-fu to handle the network side of the house, and having administrated multiple PBXs for decade-plus, I've got the concepts down. No plans to build databases for things like directories, etc... I'm not greatly confident in those skills, and to date, haven't found anything that really stands out that would make me require that. You may think otherwise, so please chime in. I say that, but at the same time I recognize I may require a GUI interface once fully deployed to allow lower-skilled people to follow the motions to complete simple moves, adds, and changes. I'm fighting the uphill battle that is the GUI is new, CLI is old mentality. System will use G.722 for VoIP Phones. So there's the groundwork. Here's the hardware plan. Plan is to build my own servers following industry standards (ATX) and using industry standard equipment. Why? Spares? Whether redundant or not, I will still have spares for the most common elements on the shelf so equipment can be returned to service as quickly as possible. This will also allow me to be comfortable with more basic server configurations and help keep cost down. For example, Servers with single power supplies vs. dual. Also, components will be standardized for all equipment to aid in supply requirements. First the layout. 2-servers acting as gateways. Each handling 2 PRIs for outside trunks. They'll also handle the analog ports. Failover will be in the form of degraded trunk access if one should fail, but the second will be able to support services in degraded fashion. 2-servers acting as VoIP PBX. A primary and a spare. Meaning one will be capable of handling the load of the entire system, and the other will pickup when the other dies, an active/passive cluster. Will also take care of voicemail. Use of heartbeat, pacemaker, etc... etc
[asterisk-users] I need a second opinion on a new phone system deployment
Howdy All, They say opinions are like belly buttons, everybody has one. (that's the clean version of the saying). So I'm asking for yours. I hope you see it as a fun exercise. I'm designing a phone system from the ground up. Will be about 1000-1300 seats mixed 80/20 VoIP/Analog. 58-acre campus environment with 23 buildings. Userbase is emergency services organization, 24/7/365 operation. Down time is not an option, but blips are acceptable. Repair time is immediate. We need failover for the failover essentially. However, money is a major factor, so I have to do it all for nothing. So here's what I'm thinking. Please throw in your 2 cents. Network will be separate for phones. Fiber infrastructure available between buildings as well as copper. Internet access will be limited to a single administrative console on a temporary basis, and then only when remote 3rd party support is required. Access for 3rd party support will be supervised through remote access tools such as VNC, GoToMeeting, etc... etc... System will have zero access to local data network. This means all ancillary support servers such as DHCP, DNS, NTP, FTP, etc...etc... will be specific to the phone system. Yes, I know some responders at this time will become fixated on me gaining this connectivity. It ain't gonna happen. It's not an option. Period, end of story. These are the parameters I must work within. Trying to fix that will be a non-starter. The phone system will upgrade an existing TDM-based system. Mitel SX2000 with NuPoint Voicemail. This will not be a dump-trunk replacement. I expect at least a one to two-year transition, meaning we will have time to find problems, work bugs, and learn over time, with minimized impacts. It also means we'll be supporting two systems for some time. PBX is 97% serving your basic phone on the desk. Nothing special. Customers expect the usual list of features. There will be a goodly number of hints required for BLF on maybe 150 phones. There is one office of about 30 phones in a call-center environment that will need that service. They would be considered low volume (but don't tell them that). My Skills... I am not a Linux kung fu master, but I have built and managed my share of Linux servers on mutiple Linux flavors. I am a DCAA, having been through formal training, and have been playing with Asterisk for years, but always in fits and spurts and never in a live environment so I am by no means a kung fu master there either. I have started dabbling with virtualizations via XEN, but I am not comfortable enough with it to go live this first round. I can see myself implementing it in about three years once we're totally comfortable with what we have, so I can then have time to get that skill sorted. I was a network engineer for the US no3. telecom for a number of years, 10-years in comm-electronics in the military before that. Telecom my entire career. I've got the kung-fu to handle the network side of the house, and having administrated multiple PBXs for decade-plus, I've got the concepts down. No plans to build databases for things like directories, etc... I'm not greatly confident in those skills, and to date, haven't found anything that really stands out that would make me require that. You may think otherwise, so please chime in. I say that, but at the same time I recognize I may require a GUI interface once fully deployed to allow lower-skilled people to follow the motions to complete simple moves, adds, and changes. I'm fighting the uphill battle that is the GUI is new, CLI is old mentality. System will use G.722 for VoIP Phones. So there's the groundwork. Here's the hardware plan. Plan is to build my own servers following industry standards (ATX) and using industry standard equipment. Why? Spares? Whether redundant or not, I will still have spares for the most common elements on the shelf so equipment can be returned to service as quickly as possible. This will also allow me to be comfortable with more basic server configurations and help keep cost down. For example, Servers with single power supplies vs. dual. Also, components will be standardized for all equipment to aid in supply requirements. First the layout. 2-servers acting as gateways. Each handling 2 PRIs for outside trunks. They'll also handle the analog ports. Failover will be in the form of degraded trunk access if one should fail, but the second will be able to support services in degraded fashion. 2-servers acting as VoIP PBX. A primary and a spare. Meaning one will be capable of handling the load of the entire system, and the other will pickup when the other dies, an active/passive cluster. Will also take care of voicemail. Use of heartbeat, pacemaker, etc... etc... 2-servers for support services. DNS, DHCP, FTP, NTP, etc... etc...Basically, everything the phones need to run plus system monitoring via something like Nagios. 1-Desktop for
Re: [asterisk-users] 911 multple-alert question
Thanks for the response. You gave me some ideas I didn't think of such
as sending a text message to the on-call security person's cell phone.
However, while I know I can get the 911 call to call other phones, I
also need location data. I know there are ways to do it, but I don't
have the kung-fu for things like databases, and am wondering if there's
something simple in Asterisk like a flat file used to correlate phone
number and location. Then, there's the part of how to get that
additional data to display on a phone. To throw a wrench in it, I don't
want local security to answer, just to be alerted the call is going on
so they can be the first on scene and make themselves available to
direct emergency personnel when they arrive.
Thanks Again!
On 6/12/2012 11:21 AM, Carlos Alvarez wrote:
On Tue, Jun 12, 2012 at 9:44 AM, Carlos Chavez
cur...@telecomabmex.com mailto:cur...@telecomabmex.com wrote:
This is as easy as running an AGI on your 911 rule to do
whatever you
want. The AGI can dial multiple phones, send emails, page you, etc.
Even without the AGI you can do many things from the dialplan.
Here's one example of non-AGI notification:
exten = s,n,System(/usr/sbin/sendEmail -t car...@televolve.com
mailto:car...@televolve.com -f swit...@televolve.com
mailto:swit...@televolve.com -u 911 call was placed -m 911 call
from ${CDR(accountcode)} ${CALLERID(num)} - ${CALLERID(name)} via
switch-1.televolve.com http://switch-1.televolve.com to
${911PROVIDER}.)
This of course requires the sendEmail program, which is hugely useful
throughout our systems for many purposes.
--
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] 911 multple-alert question
You're absolutely correct. The 911 system maintenance is a PITA. Our current PBX routes all 911 calls to an box we call the Proctor Box. That box does a couple things. First, it assigns an ANI to a number. So for example, 3-digit extensions get a real 10-digit numbers assigned. It then routes that 911 call out a dedicated CAMA trunk to the PSAP. Additionally , we pay a third party company to upload our location data to the ALI database, so when our provided ANI hits the PSAP, they can pull proper location info out of the ALI database. The other thing this box does is feed a couple local display terminals. These terminals alarm and display local information from it's own internal database during a 911 call for local security folks. In the meantime, I have the PBX itself which contains its own telephone directory with location and department information, and it's own Emergency Services ID. It will use this ID if the CAMA trunks are out of service and the phone system decides to route the 911 call out a normal PRI. This data we try to use as a baseline for the 911 data. So we currently have multiple databases I have to keep 100% in sync, with a 1000+ set campus with people moving constantly amongst the 25+ buildings. It's a nightmare. Basically, as we migrate to Asterisk, I need to figure out if I can replicate our current functionality. The preference is to come up with something much nicer than a half-dozen data points that are never in synch. If I can't find a solution in Asterisk, then I'm stuck using something like we already have. ...and yes, as you have imagined, there have been a few FUs in the past. We've dodged a number of bullets. I'm hoping I can resolve this when we migrate. I will look into the local channel variable in the sip.conf. That sounds promising. If I populate that data, how does it make it to the display of a phone on campus? I guess that's a piece I haven't either read or been able to wrap my head around yet. Thanks! On 6/12/2012 5:50 PM, Steve Edwards wrote: On Tue, 12 Jun 2012, Nunya Biznatch wrote: I also need location data. I know there are ways to do it, but I don't have the kung-fu for things like databases, and am wondering if there's something simple in Asterisk like a flat file used to correlate phone number and location. 1) The Asterisk database? You can access it with dialplan applications. 2) Set the location as a channel variable in sip.conf? (Keeps all of the 'phone specific' stuff in one place. Just being a bit paranoid, but fiddling with 911 calls always makes me nervous. I'd dial the 'real' 911 call over a copper pair first and then after (or in parallel) do all your kewl stuff. I'm sure there will be a lawyer somewhere out there just itching to sue when the 'real' 911 call doesn't happen because of some minor FU on your part. What's your legal exposure if your location data is wrong? I think I'd want a letter absolving me of liability signed by the CEO in my back pocket. -- _ -- 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
[asterisk-users] 911 multple-alert question
Can you set up asterisk so when a 911 call is placed, in addition to the call out to the PSAP, it also alerts multiple other phones on the switch and will display detailed information. Such as alerting a receptionist or security guard there is a 911 call elsewhere in the building and the location of that call within the building? If so, how? Thanks in advance for the help... -- _ -- 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] Replacing PBX with Asterisk, need feedback on my new architecture.
Finally, I'll have a Windows-based workstation that will be used to remote into all the services, for administration, etc... Why? Unfortunately, the existing PBX Administration Software only works on WinBloze. I'm stuck with it until I can decommission it. I need to plan to use FreePBX on all Asterisk Servers, but I don't intend to install it until I'm in regular MAC maintenance mode. It is ashame you are going this far with your setup to rely on FreePBX. For something this complex, you are setting your self up for some heartache. It is my intention to do everything from the command line. However, there will be times when I'll have Interns coming in and doing some of the MAC activities, and I thought this might be an easier way for the day to day to get done. I've never seen it myself either, so am curious. Finally, there's the glitter factor. When my bosses come in and want a dog and pony show on the new phone system, they want to see fluffy bunnies and kittens, not the Ox that's doing the pulling. CLI = old in the minds of those that don't comprehend. -- _ -- 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
[asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.
I'm about to receive approval to design and deploy an Asterisk-based phone system for my company. I will immediately have to start writing specifications. I'm working on the hardware design and the architecture right now. I'd like a second, third, fourth, 1,000th opinion. 800 SIP phones. All will be G.722. I expect 200 concurrent calls, with 20% leaving to the outside world. There will be another 200 analog lines that will for the time being remain on the TDM PBX switch they reside on, and will be whittled down and converted to SIP as time and attrition allows. These are primarily fax machines and conference spider phones. Those are included in my 200 concurrent calls number. I'm looking to get as close to 5-9's reliability as I can, with 4-9's mandatory. Proper power filtering and backup is already available. Here's what I'm thinking for the architecture: Server 1: PRI Gateway 1 - Support 2 outside PRI trunks for local and long distance, plus a third PRI connecting to the existing TDM PBX. Server 2: PRI Gateway 2 - Support 1 PRI trunk for local and long distance with room for another, plus a second PRI connecting to the existing TDM PBX. Reason for two PRI Gateways is for redundancy and fail-over, but processor capabilities is a concern. I expect in about two years I'll be ready to decommission the TDM PBX, but will be left with about 80 Analog lines across the multiple buildings on my campus. I expect I'll end up purchasing channel banks to support the remaining analog lines, and distribute across the campus using existing copper plant. Server 3: Asterisk Master Server Server 4: Asterisk Slave Server I'm considering a clustered environment, but I believe a fail-over solution would be easier to implement in the short term. This means each system needs to handle all traffic by itself. These servers will be used for Asterisk and Voice-mail. Conferencing will be enabled, but I'm not considering it in the build. If I see conferencing becoming a factor, I will build another server and offload that service. Server 5: Boot Server - DHCP, RADIUS, SNTP, DNS, LDAP, FTP, HTTPS, SNMP, etc... This service will provide the phone network all the basic services. This is a stand-alone phone network primarily because it would be too costly to upgrade the entire data network to support both voice and data. The phone network will not initially have Internet Access. This server will be the server all the phones talk to for pulling their configs. I'm considering a second Boot Server for redundancy, but since the phones should store their configs, I'm not seeing this as horribly critical. Am I smoking something? Finally, I'll have a Windows-based workstation that will be used to remote into all the services, for administration, etc... I need to plan to use FreePBX on all Asterisk Servers, but I don't intend to install it until I'm in regular MAC maintenance mode. I have no plans at this time to build out any databases. I just plan to use whatever Asterisk has. If it ever comes to that, I would make those separate servers as well. My goal is to build Asterisk Servers and PRI Gateways capable of supporting 150% of what I anticipate, which would come out to 300 concurrent calls. Again, all phones will use G.722. The PRI Gateway servers will do the heavy lifting of converting G.711 traffic from the PRIs to G722, and connect to the Asterisk Servers via IAX2 trunk. It's my intention to build each server myself with high-quality off the shelf components. I'd like all servers to be as close to identical as possible, as I intend to keep spares on hand to facilitate quick repair and minimize downtime. I'm considering RAID 1 + 0 (mirrored and stripped drives) for all servers. I am considering dual redundant power supplies. For a processor, I'm currently looking at the i7-3770K @ 3.5GHz or very similar. Its Passmark compares to the Xeon E5-2630 @ 2.3GHz, but is half the price. I have no idea what amount of memory to consider, so I am thinking 8GB per machine. PCI-E is what I plan for all the cards. Debian is the Linux flavor A new network will be deployed using PoE layer-2 managed switches. Battery backup capable of providing 8 hours will be installed as required. There will be multiple VLANs in the network as I have multiple dissimilar offices I need to keep separated from each other. We will also have 802.11 SIP phones, and will be deploying a campus-wide WiFi network used only by the phone system. Yes, I crunched the numbers. This will be significantly cheaper than upgrading the entire existing data network to support the new phone system. ...and to be quite honest, I don't trust our network folks, and know adding that layer of bureaucracy will only negatively impact the customer experience. I was a network engineer for a top-three telecom company for many years, so I do have a point of reference to make those statements. ...yes, I am one
Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.
Thanks for the info. It got me digging deeper. I definitely don't want to screw this one up, but I've got to pinch pennies to get this done, so don't want to buy anything that would just be nice to have. ...but if I have to get it, that's what I'll do. Have any of you seen this? ftp://download.intel.com/design/intarch/PAPERS/318862.pdf It's a whitepaper from Intel where they load tested Asterisk on various Intel Processors. They were trying to show the benefit of compiling Asterisk using their compiler vs. gcc. It's from January 2008. They used Astertest as the test base. With a dual Xeon 5335 @ 2GHz (dual quad cores), and using a gcc compiled Asterisk, they were able to process 673 concurrent calls with GSM to iLBC transcoding and 552 calls with GSM to Speex transcoding. Looking at http://cpubenchmark.net, I see a dual Xeon 5335 @ 2GHz has a Passmark score of 5,095. A more modern single E5-2630 processor has more than double the score at 10,401. ...and those results were with whatever version of Asterisk was out and about in January 2008. Would it be 1.4? From what I read here http://www.voip-info.org/wiki/view/Asterisk+dimensioning, Asterisk 1.6 is 3-4 times better in performance than 1.4, and 1.10 is 2-3 times faster than 1.8. Also, keeping in mind while yes I have 800 SIP phones, only 200 will be active concurrently at peak times based on current call traffic data, and I'm adding 50% to cover myself and looking to build to support 300 concurrent calls. Finally, throw in the fact the main Asterisk Server will not be doing any transcoding. The only transcoding will be in the PRI Gateway server, and with 3 PRI's, I only need the power to transcode 69 concurrent calls from G.711 to G.722. The next concern is the raw number of actively registered phones. I guess this is something I don't understand what the repercussions are, and I know the unknown is always what bites you. What happens? I wouldn't think that's a lot of open port traffic to worry about? Thanks Again? On 5/6/2012 3:19 PM, Mitul Limbani wrote: For 100% High Availibility and Hot Failover, I would recommend one of those Red-fone Fonebridges. Also getting 800 Phones all register on single server is crazy, add a SIP proxy to distribute load evenly between 2 Ast boxes. For Wireless you might consider using DECT phones from Snom instead of std 802.11 based wifi phones. Giving QoS on wifi is a big pain. Hope that helps, Regards, Mitul Limbani Enterux Solutions On May 6, 2012 11:34 PM, Nunya Biznatch aster...@ihearbanjos.com mailto:aster...@ihearbanjos.com wrote: I'm about to receive approval to design and deploy an Asterisk-based phone system for my company. I will immediately have to start writing specifications. I'm working on the hardware design and the architecture right now. I'd like a second, third, fourth, 1,000th opinion. 800 SIP phones. All will be G.722. I expect 200 concurrent calls, with 20% leaving to the outside world. There will be another 200 analog lines that will for the time being remain on the TDM PBX switch they reside on, and will be whittled down and converted to SIP as time and attrition allows. These are primarily fax machines and conference spider phones. Those are included in my 200 concurrent calls number. I'm looking to get as close to 5-9's reliability as I can, with 4-9's mandatory. Proper power filtering and backup is already available. Here's what I'm thinking for the architecture: Server 1: PRI Gateway 1 - Support 2 outside PRI trunks for local and long distance, plus a third PRI connecting to the existing TDM PBX. Server 2: PRI Gateway 2 - Support 1 PRI trunk for local and long distance with room for another, plus a second PRI connecting to the existing TDM PBX. Reason for two PRI Gateways is for redundancy and fail-over, but processor capabilities is a concern. I expect in about two years I'll be ready to decommission the TDM PBX, but will be left with about 80 Analog lines across the multiple buildings on my campus. I expect I'll end up purchasing channel banks to support the remaining analog lines, and distribute across the campus using existing copper plant. Server 3: Asterisk Master Server Server 4: Asterisk Slave Server I'm considering a clustered environment, but I believe a fail-over solution would be easier to implement in the short term. This means each system needs to handle all traffic by itself. These servers will be used for Asterisk and Voice-mail. Conferencing will be enabled, but I'm not considering it in the build. If I see conferencing becoming a factor, I will build another server and offload that service. Server 5: Boot Server - DHCP, RADIUS, SNTP, DNS, LDAP, FTP, HTTPS, SNMP, etc... This service will provide the phone network all the basic services
[asterisk-users] Digium FXS specifications and limits Question
Howdy All, I'm considering Asterisk / Digium as a replacement to my existing phone switch. I need to continue to be able to push analog lines between multiple buildings in a campus environment. The Digium Analog 410 Card manual states it's not recommended to go beyond 1500 feet distance for an FXS card, and no line should leave the building or be bundled. The 2400 Series Manual does not have this same notice. Should it? My potential application will be pushing all of 1500 feet and maybe a teeny bit more. Analog lines will be bundled into PE-89 type direct-burial rated cable in capacities ranging from 100-pair to 500-pair. All cable pairs are 24AWG. All cable is privately owned by the organization. All cable is professionally terminated on each end to grounded building entrance protection with lightning blocks / surge protection. This infrastructure has been in place and working on an existing old school digital switch with port cards that don't have lightning or surge suppression built in, just like the Digium cards. Has anyone run a similar configuration and can speak for or against such an idear? Thanks for the help and input, Jason -- _ -- 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
