[Asterisk-Users] Looking for an IAX(2) or SIP DID provider for LA, Orlando and Chicago areas.
I am once again looking for someone that can provide a block of DIDs in Chicagoland, LA and Orlando with a pricing model that aligns with ISDN PRI service. I would like to pay a monthly fee per simultaneous call (equivalent to a trunk) with blocks of DID numbers priced at around $0.10 or so (what I would pay for those DIDs as a retail customer in SBC territory). Thanks in advance. /carmi ___ 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] Small PBX to VoIP transition questions
On 21 Dec, 2004, at 13:54, [EMAIL PROTECTED] wrote: This is for a small business (restaurant and catering). We want to move from POTS to VoIP to save on the phone bill. Currently we use four lines for voice + one fax going into a Lucent Partner system PBX. Right now I'm considering two alternatives, both powered by * and a SIP/IAX wholesaler (any recommendations on that for someone who can do LNP on DIDs in Boston, MA, USA area are gladly accepted). Are you expecting to save on usage charges or on line costs? Are most of your calls incoming or outgoing? Alternative One) Use a Digium card in the * box to drive the PSTN lines going into the Lucent system. This is the simplest and cheapest alternative, as it leaves all the current phone equipment in place. From what I've read I think it shouldn't be a problem to get incoming calls on the VoIP to hunt through the FXS interfaces. If you want to maintain the feel of a Partner (placing calls on hold and picking up in other places, etc.) this is the only way to go. Asterisk does not support this functionality, and based on discussions on here, probably will not do so anytime soon. Alternative Two) All VoIP: buy new phones (probably Cisco or Polycom IP phones), a PoE switch and some ATAs. My hesitations in this area (aside from the cost) are mimicking the functionality of the partner system. Because this is a restaurant environment, there are only three phones that will be used as "Office" phones -- the rest are floor phones. In the partner system these are the cheapest phones offered: 4 button, no display. On these floor phones, the four buttons are just used as line buttons. Incoming calls always ring all phones. A manager can answer a call on the floor, put it on hold and return to the office. This just cannot be done with Asterisk. Instead, you would need to use call park and pick up and retrain all your staff. It also means that friends clues (like a blinking light showing a call on hold) go away. Incoming calls for employees are put on hold and the page feature is used to let them know (i.e. "Bob, call for you on line three"). It wouldn't really work to transfer the call to a specific extension, since the workers move around and need to be able to pickup an arbitrary line from anywhere there is a phone. All the lower end IP phones I look at say they have two "lines" ... I'd really like to use cheap(er) phones on the floor: they get abused and gross and don't need to do much (except be four line phones) -- and need to be replaced more frequently than "normal". I'd like to hear other's ideas on how to implement this system or if others have implemented VoIP with "floor" phones in a restaurant, warehouse, etc... Finally) Would I be foolish to try to send / receive fax over VoIP by plugging the fax server into ATAs and using a zero compression codec? We send about 150 faxes daily and receive a couple as well...having to keep more than one analog line around for FAX would defeat the cost saving motivation behind all this anyway (we can theoretically use up to three of our five lines for FAX at a time). Internet Fax is so overpriced its absurd (unless someone knows of a company doing it for around $0.02 a page, which is how much the phone call costs to send a one page fax). Do you have 3 fax machines? If not, a single analog fax line will solve your problem. It just would not be shared with anything, so there would be no contention issue. Thanks for making it through my long post...I'd by happy to get any answers about any part of what I mentioned above! ~Adam ___ 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] Re: Recommendations for full featured phones
Are you replacing a Merlin Legend (hybrid PBX/key system) or a Merlin 4/10, 8/20 low end key system? You should be aware that in its current form, Asterisk does not support shared extensions something commonly used in most key environments. /carmi On Dec 6, 2004, at 9:37 AM, Pavel Jezek wrote: look at: http://netphone.intracom.gr/english.htm we have order this meanwhile for lab testing, so I would be able to refer for about a month... PJ - Original Message - From: Sean Cook Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Monday, December 06, 2004 12:19 AM Subject: Recommendations for full featured phones We are considering a replacement of a legacy PBX system (merlin). I am trying to figure out which phones would be best supported with the fullest set of features. Any recommendations? Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
On Nov 25, 2004, at 12:37 PM, Gregory Junker wrote: I would like you to name one PBX that does not support this behavior? Every system from Avaya including their Definity, Merlin Legend, Merlin Magix, Partner, and their new IP based PBXes support it, as do those from Mitel, Nortel, InteCom and every other system that I have ever used. A typical example is a manager/admin setup that works as follows: Partner is not a PBX, it is a key system. Correct and the Legend/Magix are hybrid systems, but all of Avaya's systems including their current PBX systems (Definity) and their previous PBX versions (System 75 and System 85) support this functionality. The example that I gave comes directly from a company that uses a Definity. The Definity PBX does not directly provide key functionality. Then by your definition, this must not be key functionality as it is supported on the Definity (again, anyone that is interested in visiting several companies in the SF bay area, Chicagoland, Los Angeles or Orlando to see it in action, let me know and I will happily arrange a demo). I can't speak to Merlin, not having used it myself. I use a Merlin Legend as my primary phone system now and I can say that it supports this in just the same way as the Definity does. Again, note that I am not asking to display trunk status, just extension status, and to allow a user to place a call on hold on one phone and pick it up on another (that has that shared extension). That said, Asterisk is a PBX like Definity, and should not support this. Again, Definity already supports this, so why should Asterisk not do so? A FEP for Asterisk, that duplicates the functionality of a key system, should be developed, if it's in high enough demand. Like I said before, I am happy to spearhead the project development if anyone is interested. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
On Nov 20, 2004, at 11:05 PM, Gregory Junker wrote:
Most customers don't want to be in a new era. They want something
they are
accustomed to. I don't need any more impediments to making money than
I've
already got. So if the customer wants a busy lamp, I am going to do my
best to give it to them.
I agree. This is why engineers do not make good salesmen.
It is also why engineers make poor product marketing managers. While
there maybe many more interesting and flashy solutions that offer much
more power ("We could display current call duration and average call
duration over 1 hour, 1 day, 1 week and one month next to each user's
name allowing a receptionist to tell a caller how long an average wait
time might be.") they are often not what a product's user want or need.
On Nov 21, 2004, at 2:49 AM, Peter Svensson wrote:
On Sat, 20 Nov 2004, Brian Roy wrote:
I would look at putting a dual monitor on her desk. You can pick up a
15" flat panel and a video card for about the same cost as the SNOM.
Not to mention, you get quite a bit more benifite from the FOP
controls than you do busy lamp fields. It's a a new era here folks.
Asterisk is not your dad's pbx.
Most people here seem to miss the point that a dedicated hard
interface is
a lot easier to use than any computer interface.
...
You should always design an interface around a human being. A hard
interface with a light and a button per extension and so on is really a
very good interface. We software pople tent to forget the value of a
proper hardware solution.
Peter
I could not agree more. I think it would be great to have some other
options (like an embeded-FOP appliance), but for many basic situations
(manager/admin as one simple example) lights on a phone are hard to
beat.
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Re: [Asterisk-Users] SIP Phones-Receptionist Setup
On Nov 21, 2004, at 11:15 AM, Wayne Sheppard wrote:
Tracy R Reed wrote:
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
I'm not surprised. Asterisk is a PBX, not a key system or a hybrid
system. The kind of functionality that is being described here is one
or both of those 'other' beasts. Now I'm not saying that this wouldn't
be nice, or even a long term requirement if you really want to open
the entire SME market, but it's not typical PBX behavior.
I would like you to name one PBX that does not support this behavior?
Every system from Avaya including their Definity, Merlin Legend, Merlin
Magix, Partner, and their new IP based PBXes support it, as do those
from Mitel, Nortel, InteCom and every other system that I have ever
used. A typical example is a manager/admin setup that works as follows:
Sarah a manager has a phone on her desk with call appearances for her
main number (x-3123).
She also has a phone on her office conference table with its own
number (x-3302) but also with shared call appearances for her main
number (x-3123).
She shares a conference room with Ed, John, Steve, Susan and Simon.
All their phone numbers have shared call appearances that conference
room's phone.
Molly (Sarah's administrative assistant) has a phone with shared call
appearances for Sarah, Ed and Susan (two other Executive Team members
for whom she provides shared coverage with Wendy and Lisa).
When a call comes in for Sarah on x-3123, Molly can answer it, and just
by looking at those little red and green lights on her phone she can
tell if Sarah is on a call or not. She can then place this call on hold
(not park it, just hit that red hold button) and call Sarah announcing
this call.
Sarah can answer this call just by pressing that button next to the
flashing light (indicating a call on hold) and picking up her phone.
She does not have to use call pick up. She can also pick this call up
on her office conference table, or in the Executive Team's conference
room in exactly the same way, not needing to understand or know
anything else ("press the button with my name on it next to the
blinking green led").
All of this was done using a PBX (an Avaya Definity), never using call
pickup, or an operator console (just a standard 28 button phone for
Molly, Wendy and the Executive Team conference room, and a standard 10
button phone for Sarah, Steve, Ed, John, and Simon). This is a real
example at a real company, not just something made up as a straw man.
If you want to see examples of this, I would be happy to take you to
the Math Department at University of Illinois (Nortel), Sony Pictures
Imageworks (Avaya) or Argonne National Laboratory's Energy and
Environmental Systems group (InteCom).
In fact, if you start looking at *all* the differences in
functionality, (i.e. call announce, hands free answer-back,
hold/pickup scenarios, etc.) it *may* be easier to have a different
product stream that is targeting this sort of thing. Of course that's
easy to say, but hard to do given the number of developers that are
actually working/contributing to * on a regular basis.
I would still like to understand how adding any of these features (even
if they were not already available on almost every PBX system sold
today), would comprise Asterisk's "PBXness" in some way that would hurt
its adoption.
This isn't unique to *, it's the battle that every PBX vendor fights
at least internally with product management.
Yes, but every other PBX vendor has adopted this functionality, while
Asterisk has not.
How to be all things to all people and still have some level of
control over the product development and support streams. I guess what
I'm ultimately pointing to is the need to pre-qualify a prospect
before one makes a sales proposal.
This "religious" argument ("We cannot do that because it is
unPBX-like.") seems to also miss another important factor. While large
and small organizations use this functionality, a system is almost
unusable for a small office without it (see how it is used in every
small store or company with a Merlin Legend or Magix system for
example). I am fairly convinced that smaller offices are better
candidates to adopt Asterisk than are fortune 500 companies. Not having
these features makes Asterisk much less likely to be deployed in those
environments. While Pingtel's open source sipXchange is not quite ready
(still a month or two off from what I have seen), it is getting quite
close. I think seeding this whole market segment to them is not the
best plan.
If there are certain aspects of PBX vs. Key System that they can't
metabolize, or aren't willing to make the user training investment,
then sell them what they will can rather than try to pound a square
peg into their round hole. Does this limit the market for *? Sure
does. But
[Asterisk-Users] Using Asterisk as a replacement for a Merlin Legend.
I was just about to replace 2 Merlin Legend systems (one in my house and one in my Parent's house) with * systems. I have 2 beefy linux systems ready to go and have enough Cisco 7960 and 7970 phones to replace all my MLX-10D, MLX-16D and MLX-20 phones. Then I discovered that * seems to be lacking a critical feature for replacing a legend (or any standard PBX) and that is an ability for a user to see if another user is on his or her phone or if a trunk is in use. I am not sure if this is where I should address this issue (or if it should be send to a developers list), but is there any work underway to support this? In other words, what I would like to see is if I have a trunk that appears on multiple phones (my house's main line for example), I want a visual indication on every phone that it is in use if someone on another phone is using it. On a related front, I would like to be able to put a call on hold in one location and be able to pick it up somewhere else with out having to know anything about where it was placed on hold (difference between call park and call hold). If this is available today, it is not listed as a feature on *'s website, nor can I figure out how to make it happen from *'s documentation. To be clear, Asterisk is an amazing product and offers a great many features that I am looking forward to using. I am just trying to figure out when I will be able to use it to replace my PBXes. Thanks in advance. /carmi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comparisons between * and sipXpbx (PingTel's open source product)
Has anyone compared * to sipXpbx? From a cursory look, this open source version of PingTel's PBX has many features that make it more suitable as a replacement for a traditional PBX, including the ability for users to tell if a phone/trunk is in use. What I am trying to figure out is what I'd give up using sipX instead of * (and vice versa). /carmi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Has anyone tried using a Sipura-3000 as an FXO device for *?
I am most interested in using it for incoming calls. Have you tried that yet? /carmi On Jul 27, 2004, at 5:30 PM, Rich Adamson wrote: I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. In the process of doing that now. Simple / prelim implementation: Each of the three ports (eg, fxs, fxo, cat5) are treated as separate interfaces, and one can configure fxo -> *, fxs -> *, ring-through from fxo -> fxs, * g/w functions to the pstn, etc. There seems to be a ton of functionality in the box and those functions are mostly limited by your imagination (and how well one can read and comprehend). Configurable from a web interface, however there are a ton of options that aren't very clear without digging deep into their newly released admin manual (called a user guide on their site). The manual seems to have been written for the 1000/2000 with additional chapters/sections oriented to the 3000. (Sort of rush to print.) The fxo and fxs interfaces can be configured to register separately with *, making both very addressable, etc. Like *, it also has an internal dialplan, however understanding the various interactions requires some experimentation, as each of the interfaces seem to be considered a "gateway", and part of the dialplan directs calls to gw0, gw1, gw2 (etc) which correspond to physical interfaces in most cases. The box was truly targeted for the residential user where existing phones interface on one side, the pstn line on the other side, and the default call is sent to the voip interface. Disconnected (or failed) ethernet results in a relay flipping, tying the fxs directly to the fxo. Same with power failure. Nice. So, properly configured, it appears to be a very nice box that would allow * to sit in the middle, but still provide excellent fail-over capabilities when unusual events occur. For small installations, it makes handling US 911 calls extremely easy as that can be made part of the internal dialplan. Initial tests did not show any signs of echo, very good volume and audio quality, and would probably be a good choice for small quantities of pstn lines (particularily soho and residential users). The only downside I've seen thus far (not much experience as yet) is that * calls to the pstn line are cut through immediately, so one hears the initial dialtone from the pstn and the sending of the dtmf tones on all outgoing calls. Kind of annoying, but there might be some config option to handle it; I've just not found it as yet. (If anyone knows how to handle that, sure would appreciate a suggestion.) Thus far, I'd give the box at least an A-, and will likely move higher with a little more experience. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?
I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. /carmi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comparing * with SIPxchange
Has anyone here looked at SIPxchange or done any comparison between it and *? They seem to have a pretty full-featured open source product. Are there major advantages for either one? /carmi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Having * show lines in use on hard phones.
I am trying to figure out is * can be configured so that users would have a similar experience to what they receive on a legacy PBX. For example, on a merlin legend (or similar PBX), I have phones that have shared call appearances so that my assistant can answer my calls or see that I am on the phone, or so that I can have one phone on my desk and one at my conference table. This means that if (312) 221-1212 appears on 3 phones and is in use, all will indicate that. Can * be configured so that it works that way with any hard phones? Can Cisco 7960 and 7970's be configured that way? /carmi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Large Enterprises using asterisk
I want to clarify how this works now on *. First, on a legacy PBX, like a merlin legend, I have phones that have shared call appearances so that my assistant can answer my calls or see that I am on the phone, or so that I can have one phone on my desk and one at my conference table. This means that if (312) 221-1212 appears on 3 phones and is in use, all will indicate that. Can any hard phones (like Cisco 7960's) be configured that way? /carmi On Jul 23, 2004, at 7:43 AM, Robinson Tim-W10277 wrote: It is the hard phones that need this before Asterisk is a salable solution to small/medium businesses. What sells the system is the phones and the flashing lights. As most users already have a legacy system with a real BLF etc, until Asterisk has hard phones that have all those features it will be a tricky sale. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Hoffmeyer Sent: 23 July 2004 15:00 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Large Enterprises using asterisk PS: If already existing soft (and/or hard) phones have more of this functionality - please let me know. WAMi and other gui interfaces already support this. http://www.voip-info.org/wiki-Asterisk+WAMI We are starting work on WAMi 2.0, and I am trying to make the source available for everyone as quickly as possible. It's not a matter of the source for WAMi being open. Rather, it's just a matter of having the time to make the WAMi source code available and having the structure setup to support bugs, maintenance, and contributions. J.Christian Hoffmeyer Asterisk Solutions Group, Inc. Huntsville, AL (o)256.705.0265 (c)256.655.0321 (fax) 256.705.0280 (tf)877.ASGI.4.ME (iax) 700.ASGI.4.ME Ask me about Asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
SBC in charges $.10 a number in metro Chicago (just had to Price a PRI there). /carmi On Jul 20, 2004, at 3:12 PM, George Pajari wrote: As an example, they purchase a PRI from either an ILEC or a CLEC for between $100 and $1000 (depending on distance and market) giving them 23 voice channels and as many numbers as they want (again, numbers cost them at most between $0.01 and $0.10). Pray tell where one can purchase DIDs in quantities less than an entire NXX for 0.01 to 0.10. We're paying between CA$1.00-$2.00 ea and some of our customers in PacBell territory have indicated their pricing is comparable. g. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
What markets are you targeting? Do you have any pricing yet? /carmi On 21 Jul, 2004, at 9:51, Kevin P. Fleming wrote: James H. Cloos Jr. wrote: The demand exists; is anyone up for spulying that demand? Interesting conversation... a partner and I are setting up _exactly_ this sort of business right now, but not in the areas the OP wanted. I see a great deal of market for VOIP trunk service exactly as mentioned in this thread: multiple trunks (even some usage-based if you want to go over your "maximum"), a set of numbers (some ported via LNP, others via blocks of DID), and calls coming over those trunks just as if there was a PRI to the customer's premises. On the backend we'll do the same thing the telcos do; oversubscribe our PRI capacity at some reasonable rate. We will be targeting small businesses, averaging 20 employees and smaller, so we can use a reasonable oversubscription rate. A company selling this service to larger companies would have to have a closer match between their number of "sold" VOIP trunks and their number of PRI channels available. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
On 20 Jul, 2004, at 21:37, Steven Critchfield wrote: On Tue, 2004-07-20 at 15:58, Carmi Weinzweig wrote: Chris - In the real telephony world, one can buy a DID trunk without buying a PRI. If one wants more than about 10 trunks (depending on provider), it may be cheaper to buy a PRI instead of individual trunks. Having said that, most of these VoIP providers have their pricing model exactly backwards (they seem to only want to compete with Centrex, not with regular PBX services), in that they charge a lot for resources that are freely available and cost them little (phone numbers), but very little for scarce resources (call terminations) that cost them much more. Cost isn't a determination of scarcity. Numbers are scarce in the fact that they get assigned out to a specific entity and for a time, it is associated with that single entity. Phone lines for a VoIP provider though can be shared amongst the entire customer base. To be clear, I was not talking about global resources, just in terms of those of an VoIP provider. Adding additional channels requires adding hardware (PRI card, CSU/DSU, possibly more CPU power) and takes more network bandwidth. Adding numbers (which are plentiful right until they are no long available) incurs no other cost (well, maybe a small ordering fee). A PRI circuit should be between $35 and $50 per channel, Split amongst 3-5 customers. Of course you have to then account for the data side of the network too. Last time we discussed with our telco pricing on DIDs, it was $4/month per 20 numbers. Anything more than a couple blocks required some justification. I think it was basically to make sure we wheren't running some form of scam and moveing from number to number. So, you are paying $.20 per number, in TN. Last I checked here in Sprint Local Territory, they were $0.10. My deal with AT&T Local Services in SF (a year or two ago) they bundled some and so we ended up paying about $0.01 per number. /carmi What I would like is to be limited as to how much of a scarce resource (channels) I can use, but not be limited as to how much of a plentiful resource (numbers) I can use. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
Chris - In the real telephony world, one can buy a DID trunk without buying a PRI. If one wants more than about 10 trunks (depending on provider), it may be cheaper to buy a PRI instead of individual trunks. Having said that, most of these VoIP providers have their pricing model exactly backwards (they seem to only want to compete with Centrex, not with regular PBX services), in that they charge a lot for resources that are freely available and cost them little (phone numbers), but very little for scarce resources (call terminations) that cost them much more. As an example, they purchase a PRI from either an ILEC or a CLEC for between $100 and $1000 (depending on distance and market) giving them 23 voice channels and as many numbers as they want (again, numbers cost them at most between $0.01 and $0.10). They assign me a phone number (a value of $0.01 and $0.10) and let me receive as many simultaneous calls as my bandwidth allows (using these numbers every call absorbs a channel that costs between $4.35 and $43.48). What I would like is to be limited as to how much of a scarce resource (channels) I can use, but not be limited as to how much of a plentiful resource (numbers) I can use. /carmi On 20 Jul, 2004, at 14:22, Chris A. Icide wrote: On 10:41 AM 7/20/2004, Carmi Weinzweig wrote: >I want many phone numbers so that each phone in my facility has its own >phone number, but I really do not need that many simultaneous calls and >it would be cost prohibitive to pay several dollars for each phone >number. It's a different business plan. By going to a VoIP provider, you alleviate the requirement for hardwware you lease or own to terminate PRI's at multiple locations and distribute the calls to your end users. So, you aren't paying for the physical T1 and associated hardware. The VoIP providers are now incurring that cost and must recuperate it (unless they are operatiing under the '90s dot com business plans in which recuperating costs is not required - but you better be ready to turn up a new provider on a moments notice if you are using one of these). So in the past if I am understanding you, you would buy a PRI and pay some fee for the T1 itself, as well as $0.01 to $0.10 per number assigned. In this case, you want to not pay the T1 fee but still pay low per number rates. Maybe if you talked to the providers they might come to a different pricing plan for you that emulates the old way and gives you a better bang for the number? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] PRI dead in USA?
If one is using BRI primarily for voice, POTS lines while they will work are not a great replacement for many reasons: No reliable disconnect Slower call setup times Slower and less reliable number delivery (CallerID vs. ANI) Lower voice quality While no one seems to actually support it, it should be possible to support DID over BRI, which would make small BRI installations very cost effective. Fractional PRI might be something to consider as well, but it really depends on how many lines you need and what SBC's gouge rates are these days. I'd also check with the CLECs there. /carmi On 20 Jul, 2004, at 12:46, Doug Shubert wrote: Hi Scott, Local ISDN BRI service is definitely on it's way out. We recently have canceled several ISDN BRI accounts and replaced them with ADSL lines. More bandwidth and less cost. If you intend on using the lines for voice only, then FXO is the better option. If you looking to use voice&data the I would suggest 1 FXO line with ADSL over it. We believe the Digium cards with Asterisk in a small Linux box will provide a best combination of flexibility and services. Doug, Voippages.com Scott Stingel wrote: Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were "well, BRI is getting quite antiquated", and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built-in feature. Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
I am looking for a provider that will provide an equivalent of DID/DOD trunks via IAX, IAX2 or SIP using numbers in Metro Chicago (prefer Skokie), LA (prefer West Hollywood or Venice), and/or Orlando (prefer Winter Garden). If I can migrate some of my existing numbers using LNP, that would be even better, but it is not a requirement. While I know that there are several companies that will terminate VoIP number using these protocols, none offers a functional equivalent of ILEC DID service. From my ILEC, I can purchase one or more DID trunks and a block of phone numbers (usually for between $0.01 and $0.10 a number). I can receive as many calls simultaneously as I have trunks, after that callers receive a busy signal. All VoIP trunk providers that I have found, want to charge me several dollars per phone number, but will allow me unlimited incoming calls per number. I want many phone numbers so that each phone in my facility has its own phone number, but I really do not need that many simultaneous calls and it would be cost prohibitive to pay several dollars for each phone number. Thanks in advance. /carmi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
