Greetings I'm in the process of planning my first production system and wondered if those with some experience would let me know if I'm doing anything stupid or have some suggestions. This is going to be used in a manufacturing facility with about 22 phones. About 10 of which are office staff. I'm not going to implement call recording, meetme, or queues or anything fancy at this point. I'll be using Polycom 601's and 501's for the office staff and 301's for the plant phones. I've already had a few phones set up in my office to test with and I've got what I need for provisioning figured out. I'll have all the phones set to canreinvite=yes and use the transfer functions of the phone. Voice mail will be provided for office staff. Since I don't have that many phones and everything will be on the LAN I'm just going to stick with ulaw for the codec. The planned server will be an HP Proliant ML110 G3 with a 3Ghz Pentium 630 processor, 1GB of RAM, and two 80GB SATA hard drives in a RAID1 (linux software raid) configuration. I'm planning on using the on-board gigabit network controller. I'll have about 8 POTS lines (no caller id or call waiting) connected to the system. I'm planning on using a Sangoma Remora A20004D (8 FXO with on-board echo canceler). Echo is actually my biggest fear of the whole project. There won't be any faxes coming through the server. For the few analog phones that may be used I'll be using some SPA-3000's I already have on hand for FXS ports. We will have need for overhead paging eventually. This is one area I'm a little unsure of. My current off-the-cuff plan is to use a Budgetone phone with the headset jack plugged into the amp and set to auto-answer. (Saw this on the wiki). I've looked at some of the other devices on the wiki but I'm not sure how to implement them. Any advice would be appreciated. I'm also trying to decide whether I want to use Asterisk Business Edition or stick with the downloaded version. Money really isn't a big issue but I'm not sure what the pros and cons are. I know I would get a "hardened" version thats not likely to have many bugs and support from Digium, but I'm not sure what "version" of asterisk it is or what features are in the 1.2 branch that aren't in ABE or vice-versa. I'm assuming ABE is in binary form, will it even work with Sangoma hardware, is it distro sensitive? (I was going to call Digium but ran out of time this week). I think that covers it. If anyone has some tips or constructive criticism I would appreciate hearing it. Thanks! -Dave
This slashdot article may help you with the paging portion of your endevaour: http://ask.slashdot.org/article.pl?sid=06/09/18/220222&threshold=1 -brandon On 9/22/06, Dave Fullerton <dfullertasterisk@shorelinecontainer.com> wrote:> > > Greetings > > I'm in the process of planning my first production system and wondered > if those with some experience would let me know if I'm doing anything > stupid or have some suggestions. > > This is going to be used in a manufacturing facility with about 22 > phones. About 10 of which are office staff. I'm not going to implement > call recording, meetme, or queues or anything fancy at this point. I'll > be using Polycom 601's and 501's for the office staff and 301's for the > plant phones. I've already had a few phones set up in my office to test > with and I've got what I need for provisioning figured out. I'll have > all the phones set to canreinvite=yes and use the transfer functions of > the phone. Voice mail will be provided for office staff. Since I don't > have that many phones and everything will be on the LAN I'm just going > to stick with ulaw for the codec. > > The planned server will be an HP Proliant ML110 G3 with a 3Ghz Pentium > 630 processor, 1GB of RAM, and two 80GB SATA hard drives in a RAID1 > (linux software raid) configuration. I'm planning on using the on-board > gigabit network controller. > > I'll have about 8 POTS lines (no caller id or call waiting) connected to > the system. I'm planning on using a Sangoma Remora A20004D (8 FXO with > on-board echo canceler). Echo is actually my biggest fear of the whole > project. There won't be any faxes coming through the server. > > For the few analog phones that may be used I'll be using some SPA-3000's > I already have on hand for FXS ports. > > We will have need for overhead paging eventually. This is one area I'm a > little unsure of. My current off-the-cuff plan is to use a Budgetone > phone with the headset jack plugged into the amp and set to auto-answer. > (Saw this on the wiki). I've looked at some of the other devices on the > wiki but I'm not sure how to implement them. Any advice would be > appreciated. > > I'm also trying to decide whether I want to use Asterisk Business > Edition or stick with the downloaded version. Money really isn't a big > issue but I'm not sure what the pros and cons are. I know I would get a > "hardened" version thats not likely to have many bugs and support from > Digium, but I'm not sure what "version" of asterisk it is or what > features are in the 1.2 branch that aren't in ABE or vice-versa. I'm > assuming ABE is in binary form, will it even work with Sangoma hardware, > is it distro sensitive? (I was going to call Digium but ran out of time > this week). > > I think that covers it. If anyone has some tips or constructive > criticism I would appreciate hearing it. > > Thanks! > > -Dave > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Brandon Galbraith Email: brandon.galbraith@gmail.com AIM: brandong00 Voice: 630.400.6992 "A true pirate starts drinking before the sun hits the yard-arm. Yarrrr. --thelost" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060922/c14fdd8c/attachment.htm
Paging: You can also use the server's audio card for paging, if it is close to the main amp. Beware of Bogen or Valcom as they mainly make FXO paging interfaces for trunk lines. Viking makes the paging toys you would want to look at. ABE: ABE is the supported version of Asterisk, same code but someone to call on the phone. I am sure that they can help you with your card. Depending on your comfort level you can select ABE or Public. No real draw backs, one benefit is that buying from Digium will keep the developers in free pizza land for longer. :-) POTS: While more exspensive a PRI T1 would better suit you. Yes it cost more, but it has some advantages. A channel bank is a little handier than an ATA, you just plug in and test to your hearts content. Just did a setup of this size, company was quoted 60k+ from the phone company so don't be afraid to setup PRI, Channelbanks for future testing or use as you will fall way under budget. On 9/22/06, Dave Fullerton <dfullertasterisk@shorelinecontainer.com> wrote:> > Greetings > > I'm in the process of planning my first production system and wondered > if those with some experience would let me know if I'm doing anything > stupid or have some suggestions. > > This is going to be used in a manufacturing facility with about 22 > phones. About 10 of which are office staff. I'm not going to implement > call recording, meetme, or queues or anything fancy at this point. I'll > be using Polycom 601's and 501's for the office staff and 301's for the > plant phones. I've already had a few phones set up in my office to test > with and I've got what I need for provisioning figured out. I'll have > all the phones set to canreinvite=yes and use the transfer functions of > the phone. Voice mail will be provided for office staff. Since I don't > have that many phones and everything will be on the LAN I'm just going > to stick with ulaw for the codec. > > The planned server will be an HP Proliant ML110 G3 with a 3Ghz Pentium > 630 processor, 1GB of RAM, and two 80GB SATA hard drives in a RAID1 > (linux software raid) configuration. I'm planning on using the on-board > gigabit network controller. > > I'll have about 8 POTS lines (no caller id or call waiting) connected to > the system. I'm planning on using a Sangoma Remora A20004D (8 FXO with > on-board echo canceler). Echo is actually my biggest fear of the whole > project. There won't be any faxes coming through the server. > > For the few analog phones that may be used I'll be using some SPA-3000's > I already have on hand for FXS ports. > > We will have need for overhead paging eventually. This is one area I'm a > little unsure of. My current off-the-cuff plan is to use a Budgetone > phone with the headset jack plugged into the amp and set to auto-answer. > (Saw this on the wiki). I've looked at some of the other devices on the > wiki but I'm not sure how to implement them. Any advice would be > appreciated. > > I'm also trying to decide whether I want to use Asterisk Business > Edition or stick with the downloaded version. Money really isn't a big > issue but I'm not sure what the pros and cons are. I know I would get a > "hardened" version thats not likely to have many bugs and support from > Digium, but I'm not sure what "version" of asterisk it is or what > features are in the 1.2 branch that aren't in ABE or vice-versa. I'm > assuming ABE is in binary form, will it even work with Sangoma hardware, > is it distro sensitive? (I was going to call Digium but ran out of time > this week). > > I think that covers it. If anyone has some tips or constructive > criticism I would appreciate hearing it. > > Thanks! > > -Dave > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) lathama@lathama.com - lathama@gmail.com If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. ---