[asterisk-users] Analog FXO or IAX DIDS for new facility?
I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read Asterisk: The Future of Telephony and done a fair amount of googling. I am completely sold on Asterisk, and the new building's phones will be a mix of SIP handsets and softphones. I am confused about one thing: Should we be getting a block of analog circuits from the local telco (probably ATT), connected to the server's FXO cards for in-bound and out-bound POTS calls; or should we get a block of DIDS numbers from one of the plethora of providers available over the Internet, and then have our server connect POTS calls by IAX to the DIDS provider? We are unsure whether we are going to have separate numbers for everyone in the organization, or just 1 US phone number, with everyone in the org having their own extension number. That probably largely depends upon cost. We will have 75 people in the building. We have no data on call patterns or usage (because our legacy system belongs to our current facilities host), but we currently have 4 lines for 35 people and on unusual occasions they all get busy. An additional consideration is that we also have 300 other people scattered literally world-wide, and the next logical future step is to start providing VOIP links for them, as well. Thanks in advance for your advice. Any other suggestions, such as # of lines sizing info or reputable DIDS vendors (if that's the answer) are also appreciated. -- Sincerely Yours, Stephen P. Fierbaughstep...@fierbaugh.org Pioneer Bible Translatorsstephen.fierba...@pbti.org Pronounced: Fire as in hot, Bah as in humbug! John 3:16 in over 3,000 languages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog FXO or IAX DIDS for new facility?
My .02 - IAX may not be an option and is probably not a good one if it is. It requires a good bit of overhead to work reliably and well. You won't go wrong using SIP DID's, but if you use Analog FXO, I'd go for an 8 or 16 port card and make sure you get the card away from any existing IRQ's, especially the RAID one. If you went SIP DID instead of FXO, this would make putting your world-wide folks in an easier task. IMO a pretty good rule-of-thumb is that a line for every 8 folks will generally work pretty well, with a minimum of 3 lines. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Fierbaugh (PBT) Sent: Thursday, July 23, 2009 9:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Analog FXO or IAX DIDS for new facility? I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read Asterisk: The Future of Telephony and done a fair amount of googling. I am completely sold on Asterisk, and the new building's phones will be a mix of SIP handsets and softphones. I am confused about one thing: Should we be getting a block of analog circuits from the local telco (probably ATT), connected to the server's FXO cards for in-bound and out-bound POTS calls; or should we get a block of DIDS numbers from one of the plethora of providers available over the Internet, and then have our server connect POTS calls by IAX to the DIDS provider? We are unsure whether we are going to have separate numbers for everyone in the organization, or just 1 US phone number, with everyone in the org having their own extension number. That probably largely depends upon cost. We will have 75 people in the building. We have no data on call patterns or usage (because our legacy system belongs to our current facilities host), but we currently have 4 lines for 35 people and on unusual occasions they all get busy. An additional consideration is that we also have 300 other people scattered literally world-wide, and the next logical future step is to start providing VOIP links for them, as well. Thanks in advance for your advice. Any other suggestions, such as # of lines sizing info or reputable DIDS vendors (if that's the answer) are also appreciated. -- Sincerely Yours, Stephen P. Fierbaughstep...@fierbaugh.org Pioneer Bible Translatorsstephen.fierba...@pbti.org Pronounced: Fire as in hot, Bah as in humbug! John 3:16 in over 3,000 languages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog FXO or IAX DIDS for new facility?
On Thu, Jul 23, 2009 at 5:08 PM, Danny Nicholasda...@debsinc.com wrote: My .02 - IAX may not be an option and is probably not a good one if it is. It requires a good bit of overhead to work reliably and well. You won't go wrong using SIP DID's, but if you use Analog FXO, I'd go for an 8 or 16 port snip I second what Danny said, go for SIP DID, there are many good providers and you could even have local DID in different countires if that made it easier for your correspondents. There are IAX providers too , though if you have a compelling reason to use IAX. Go with a solid, long running company on the DIDs. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog FXO or IAX DIDS for new facility?
In australia, I would usually suggest a mix of E1 and SIP for calls - it doesn't cost any money to receive calls via E1, and redundancy is an old, valuable friend of mine. PaulH Stephen Fierbaugh (PBT) wrote: I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read Asterisk: The Future of Telephony and done a fair amount of googling. I am completely sold on Asterisk, and the new building's phones will be a mix of SIP handsets and softphones. I am confused about one thing: Should we be getting a block of analog circuits from the local telco (probably ATT), connected to the server's FXO cards for in-bound and out-bound POTS calls; or should we get a block of DIDS numbers from one of the plethora of providers available over the Internet, and then have our server connect POTS calls by IAX to the DIDS provider? We are unsure whether we are going to have separate numbers for everyone in the organization, or just 1 US phone number, with everyone in the org having their own extension number. That probably largely depends upon cost. We will have 75 people in the building. We have no data on call patterns or usage (because our legacy system belongs to our current facilities host), but we currently have 4 lines for 35 people and on unusual occasions they all get busy. An additional consideration is that we also have 300 other people scattered literally world-wide, and the next logical future step is to start providing VOIP links for them, as well. Thanks in advance for your advice. Any other suggestions, such as # of lines sizing info or reputable DIDS vendors (if that's the answer) are also appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
