Get the latest models of shoes on our online site - all the famous brands are available
Make sure you get the latest models of shoes at the lowest prices anywhere. You will love what you see - all the famous brands are there
Enter
Tuesday, September 30, 2008
Ferragamo shoes at factory prices
Re: [asterisk-biz] US/Canada DID Providers who expose full DID inventory
On Wed, 1 Oct 2008, Rehan Allah Wala wrote:
> I am now 35 years old, My english sucks, I was born and raised and went to school in
> Pakistan btw, Can u speak a few words from any other languages that I speak btw including
> Hindi, Urdu, Punjabi, Sindhi, Memoni btw ?
I'm sure any lists in those languages would give you the same treatment if
you typed your emails in the way you post to this list. People take
language mastery as a sign of understanding of a culture and a willingness
to learn. I've known of you Rehan for at least 2 years -- take some
classes on how to write proper English.
We all make typos, but if you want to be respected in an English-speaking
mailing list and as a business owner, you need to become much better at
composing properly typed, spelled and grammatically correct missives.
Start by spelling out "you" rather than "u" (this isn't a txt msg), not
using "btw" or other abbreviations 3 times in a run-on sentence, and by
generally not being defensive.
A rewrite of your response:
"I am 35 years old, and I have yet to master English. Unfortunately, it's
not a priority for me right now, as I'm focused on growing my bottom
line. I was born and raised in Pakistan, where I also attended school.
I am able to read, write and speak Hindi, Urdu, Punjabi, Sindhi and
Memoni fluently, but English escapes me. Please continue to bug me about
it, however, as I would love to master both written and spoken English.
I'm sure this too will improve my bottom line!"
Seriously, all kidding aside, you will do better on this list, on your
various sites, and in general you will get more respect from native English
speakers if you take the time to really learn and master the language. You
already know 5 languages -- what's a 6th?
Beckman
---------------------------------------------------------------------------
Peter Beckman Internet Guy
beckman@angryox.com http://www.angryox.com/
---------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] NPA/NXX Termination Breakout Rates
On Sep 30, 2008, at 11:21 PM, SIP wrote:
> Peter Beckman wrote:
>>
>> Also, who offers free Toll-Free termination?
>>
Just send the calls there via sip.
Mike
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] US/Canada DID Providers who expose full DID inventory
You can start by improving some things in your posting.
"u would like" and "greatful" do not indicate a strong commitment to
accessible, correct English. :-)
Rehan Allah Wala wrote:
> Hello,
>
> Sorry I do not know your name,
>
> I would be greatful if u can send us some things that u would like to have changed on the
> web site, and the problems with english that you are facing.
>
> We would love to improve them, and issue you credit on each bug that you report also.
>
> Rehan
> SuperTec.com
> Makers of DIDX.net
>
>
>>>> DIDx: API, Full DID (eww)
>>> DIDX, for a while already, no longer lets you browse the full
>>> inventory, at least through their website.
>> There are a lot of bugs on DIDx website, and the overall level of english
>> leaves a lot to be desired. On the otherhand they are still the cheapest
>> one I've found for a wide range of DIDs and I've yet to have any
>> significant issue with them.
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-biz mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>
>
> Rehan Ahmed AllahWala
> Msn/Yahoo/GoogleTalk/Email: Rehan@Rehan.com
> http://www.supertec.com/ - Internet Telephony Solutions
> Http://www.DIDX.net - DID Number Market Place.
> Don't Remember Me ? Visit http://www.Rehan.com
>
> ~~~~~~~~~~~~~~~~~~~
> "First they ignore you, then they laugh at you, then they fight you, then you win."
> By Gandhi.
>
> "Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Pakistani DID numbers $1.99 a month
Dear All,
We are happy to announce that we are able to get the rates for Pakistani DID numbers to 1.99$ a month.
This is a Special Price till we last the stock on this Order # 710891 on didx.net
We hope you enjoy them, and so will your customers.
DIDX.net
Rehan Ahmed AllahWala
Msn/Yahoo/GoogleTalk/Email: Rehan@Rehan.com
http://www.supertec.com/ - Internet Telephony Solutions
Http://www.DIDX.net - DID Number Market Place.
Don't Remember Me ? Visit http://www.Rehan.com
~~~~~~~~~~~~~~~~~~~
"First they ignore you, then they laugh at you, then they fight you, then you win."
By Gandhi.
"Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi
Re: [asterisk-biz] NPA/NXX Termination Breakout Rates
Peter,
We can terminate your toll free calls for you for free.
Michael Christiansen
Ohio Telecom
----- Original Message -----
From: "Peter Beckman" <beckman@angryox.com>
To: <asterisk-biz@lists.digium.com>
Sent: Tuesday, September 30, 2008 10:52 PM
Subject: [asterisk-biz] NPA/NXX Termination Breakout Rates
> A few months (maybe a year) ago someone posted a fairly aggressive US
> Termination pricing broken out by NPA/NXX. I'm interested to see what the
> current rates are these days if I break out my calls by NPA/NXX. I don't
> have that many minutes (100k) per month, but I'm always interested in
seeing
> what's out there.
>
> Also, who offers free Toll-Free termination?
>
> For those looking for great US/Canada termination, I should mention that
> VoIPInvite rocks. I use them currently for all my termination, and they
> are awesome.
>
> I'm looking for 2 reasons --
> 1. I want to have a backup provider for disaster planning
> 2. I want to see if I can reduce costs for the current number of
calls
> I'm handling now. I haven't asked Vijay for an NPA/NXX deck yet,
so
> Vijay, I'll email you separately. :-)
>
> PS -- What happend to Brian Fertig and Molten Telecom? Their website went
> down in February... I think that's who I was working with when I was
> considering termination providers.
>
> Beckman
> --------------------------------------------------------------------------
-
> Peter Beckman Internet
Guy
> beckman@angryox.com
http://www.angryox.com/
> --------------------------------------------------------------------------
-
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] NPA/NXX Termination Breakout Rates
Peter Beckman wrote:
> A few months (maybe a year) ago someone posted a fairly aggressive US
> Termination pricing broken out by NPA/NXX. I'm interested to see what the
> current rates are these days if I break out my calls by NPA/NXX. I don't
> have that many minutes (100k) per month, but I'm always interested in seeing
> what's out there.
>
> Also, who offers free Toll-Free termination?
>
> For those looking for great US/Canada termination, I should mention that
> VoIPInvite rocks. I use them currently for all my termination, and they
> are awesome.
>
> I'm looking for 2 reasons --
> 1. I want to have a backup provider for disaster planning
> 2. I want to see if I can reduce costs for the current number of calls
> I'm handling now. I haven't asked Vijay for an NPA/NXX deck yet, so
> Vijay, I'll email you separately. :-)
>
> PS -- What happend to Brian Fertig and Molten Telecom? Their website went
> down in February... I think that's who I was working with when I was
> considering termination providers.
>
>
> A few months (maybe a year) ago someone posted a fairly aggressive US
> Termination pricing broken out by NPA/NXX. I'm interested to see what the
> current rates are these days if I break out my calls by NPA/NXX. I don't
> have that many minutes (100k) per month, but I'm always interested in seeing
> what's out there.
>
> Also, who offers free Toll-Free termination?
>
> For those looking for great US/Canada termination, I should mention that
> VoIPInvite rocks. I use them currently for all my termination, and they
> are awesome.
>
> I'm looking for 2 reasons --
> 1. I want to have a backup provider for disaster planning
> 2. I want to see if I can reduce costs for the current number of calls
> I'm handling now. I haven't asked Vijay for an NPA/NXX deck yet, so
> Vijay, I'll email you separately. :-)
>
> PS -- What happend to Brian Fertig and Molten Telecom? Their website went
> down in February... I think that's who I was working with when I was
> considering termination providers.
>
>
Brian left Molten shortly before they imploded, I believe. I'm guessing
(pure speculation based on hearsay, rumour, and wild supposition) that
they cut all staff because they didn't have the money to keep them. And
shortly thereafter, cut their losses and closed.
We looked at VoIPInvite, but couldn't get things to work with their
systems. We support reinvites to keep minimum paths between callers that
can use them, but as soon as the reinvite came through, their hardware
cut us off. They had pretty good rates, though.
We've had excellent luck with Vitelity for US/Canada/A-Z termination
quality, but after signing their contracts, they seem to have slipped
something in that wasn't in their contract anywhere (which we're still
deciding whether or not to fight). They charge for international calls
(by that, they mean anything not US/Canada, which means the majority of
OUR calls as most of our users are not from US/Canada) based on when a
call is initiated as opposed to when it connects. Meaning, if the call
takes four rings to answer, that's another 20 seconds or so of billing.
If it takes a LONG time to answer, it could be an extra minute just
because the call took time to connect. Utterly non-standard, and against
their contract, but that's what they say they're 'required to do' from
their underlying carrier. If you bill with a minimum of 1 minute billing
(as we do), then it makes little difference in the long run. If you're
billing 6-second, it can make a huge offset in your billing.
We had RNK Telecom for a while. Excellent rates. But their deck rates
change weekly (sometimes DRASTICALLY), and are sent out in excel
spreadsheet files, making the whole thing time consuming and difficult
to automate. Their quality is also somewhat sub-par, and there were
NUMEROUS occasions when a number or entire region would be unreachable,
and after days and days of calling and trying to get a response, we'd be
told a rather terse, "Sorry. There's a problem, but we can't fix it.
Choose someone else for that route." This happened with some US/Canada
numbers, and entire countries like Norway and Italy. While their rates
were very competitive, for those reasons, I would most certainly not
recommend them.
Jay Kordic, who lurks around here, represents several carriers with
excellent rates. Excel is one of them with fantastic A-Z and US/Canada
termination and I vaguely remember they had very competitive rates and a
nicely distributed network.
PointOne does fantastic quality for US/Canada termination, but their A-Z
rates are pretty high.
There are the usual other players like Teliax, Bandwidth.com, etc. No
experience with them directly, but I hear raves aplenty about them. Not
sure about their deck rates, though.
Most of these players (if not all) offer Toll-free termination. Of
course, it's not toll-free. But that's the game. ;)
N.
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Branded shoes at affordable prices now
Hundreds of models of shoes like Gucci and Armani at highly discounted prices. Superior quality and unmatched only at a limited time offer. See our online shop now
Enter
Branded shoes at a fraction of the price, just for you
Hundreds of models of shoes like Gucci and Armani at highly discounted prices. Superior quality and unmatched only at a limited time offer. See our online shop now
Enter
[asterisk-biz] NPA/NXX Termination Breakout Rates
A few months (maybe a year) ago someone posted a fairly aggressive US
Termination pricing broken out by NPA/NXX. I'm interested to see what the
current rates are these days if I break out my calls by NPA/NXX. I don't
have that many minutes (100k) per month, but I'm always interested in seeing
what's out there.
Termination pricing broken out by NPA/NXX. I'm interested to see what the
current rates are these days if I break out my calls by NPA/NXX. I don't
have that many minutes (100k) per month, but I'm always interested in seeing
what's out there.
Also, who offers free Toll-Free termination?
For those looking for great US/Canada termination, I should mention that
VoIPInvite rocks. I use them currently for all my termination, and they
are awesome.
I'm looking for 2 reasons --
1. I want to have a backup provider for disaster planning
2. I want to see if I can reduce costs for the current number of calls
I'm handling now. I haven't asked Vijay for an NPA/NXX deck yet, so
Vijay, I'll email you separately. :-)
PS -- What happend to Brian Fertig and Molten Telecom? Their website went
down in February... I think that's who I was working with when I was
considering termination providers.
Beckman
---------------------------------------------------------------------------
Peter Beckman Internet Guy
beckman@angryox.com http://www.angryox.com/
---------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Asterisk support around the Walnut Creek, CA (Bay Area)
Would you be interested in a hosted asterisk pbx ?
If yes, You can explore ip-pabx.com from my company
It is free for upto 10 extentions, you just pay for usage of mins and phone lines.
Rehan
> Are there any local Asterisk support around the Bay Area? I am considering implementing Asterisk PBX in 2-3 months and would love to get some assistance.
>
> Thanks!
> Hin
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
Rehan Ahmed AllahWala
Msn/Yahoo/GoogleTalk/Email: Rehan@Rehan.com
http://www.supertec.com/ - Internet Telephony Solutions
Http://www.DIDX.net - DID Number Market Place.
Don't Remember Me ? Visit http://www.Rehan.com
~~~~~~~~~~~~~~~~~~~
"First they ignore you, then they laugh at you, then they fight you, then you win."
By Gandhi.
"Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi
Re: [asterisk-biz] US/Canada DID Providers who expose full DID inventory
>> DIDx: API, Full DID (eww)
>
> DIDX, for a while already, no longer lets you browse the full
> inventory, at least through their website.
>
> DIDX, for a while already, no longer lets you browse the full
> inventory, at least through their website.
There are a lot of bugs on DIDx website, and the overall level of english
leaves a lot to be desired. On the otherhand they are still the cheapest
one I've found for a wide range of DIDs and I've yet to have any
significant issue with them.
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] US/Canada DID Providers who expose full DID inventory
On Tue, Sep 30, 2008 at 2:09 PM, Peter Beckman <beckman@angryox.com> wrote:
> DIDx: API, Full DID (eww)
> DIDx: API, Full DID (eww)
DIDX, for a while already, no longer lets you browse the full
inventory, at least through their website.
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Large Enterprise integrator needed
I've been contacted by a Fortune 100 company who wants to learn more
about Enterprise-scale Asterisk installation options. They're just
starting to investigate possible solutions, but would like to hear
from a wide variety of possible contractors who could install or
assist in integrating Asterisk into their very large telephony
network.
about Enterprise-scale Asterisk installation options. They're just
starting to investigate possible solutions, but would like to hear
from a wide variety of possible contractors who could install or
assist in integrating Asterisk into their very large telephony
network.
Some basic areas of discussion that an integrator would have to provide:
- 24x7 support
- call center solution
- UC integration (mail/vm, fax, instant messaging)
- configuration tools/databases
- redundancy and failure-proofing
This may be a "one way communication," meaning that if you send me
your contact data I'll merely pass it to them and they will do their
own examination of your web site, etc. - for the purposes of this
transaction, I'm merely someone who knows how to post to the -biz
list on their behalf, and I'm betting that during the process of
integration there will be some good patches and features added by the
effort.
Please forward me:
- your name
- your email address
- your phone number
- your company website which contains clear descriptions of how you
have integrated Asterisk in the past
I will in turn relay this to my contact at this company. North
American consultancy shops preferred at this point, but having
international presence is useful.
JT
--
John Todd
jtodd@digium.com +1-256-428-6083
Asterisk Open Source Community Director
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] US/Canada DID Providers who expose full DID inventory
Hello!
Not exactly what you're looking for, but you might want to add
Future Nine to the non-immediate list. We take about 1-2 days
to provision DIDs as we order them especially for you. Pricing
is very competitive, and we have no minimums or any other such
headaches. You pay for what you buy - that's it.
For our standard pricing:
http://www.future-nine.com/plans.html
For a list of rate centers we offer with our standard pricing:
http://www.future-nine.com/rate-centers.txt
We also support and have a very good success ratio with number
porting in these rate centers.
If you have any questions, feel free to drop me a line.
Thanks!
--
Nitzan Kon, CEO
Future Nine Corporation
www.future-nine.com
--- On Tue, 9/30/08, Peter Beckman <beckman@angryox.com> wrote:
> From: Peter Beckman <beckman@angryox.com>
> Subject: [asterisk-biz] US/Canada DID Providers who expose full DID inventory
> To: asterisk-biz@lists.digium.com
> Date: Tuesday, September 30, 2008, 2:09 PM
> I'm trying to find more US and Canada (NANPA generally)
> Wholesale DID
> origination providers who, in order of importance to me:
>
> 1 expose full DID for inventory
> 2 immediate provisioning of in-stock DIDs
> 3 have good quality, reliability and are responsive
> 4 Have an API
> 5 offer per-minute pricing, competitive pricing
> 6 prepaid
>
> Here's the providers I know about already
>
> Vitelity: API, Full DID
> Junction: API, NPANXX only
> Les.net: No API, Full DID
> DIDww: API, NPA only
> DIDx: API, Full DID (eww)
>
> Providers I am aware of, but would love comments:
>
> Unlimited Net: API, don't know about exposure
> VoicePulse: API, Full DID (I think)
> Voice Network: Dunno
>
> Providers that don't expose DID:
>
> Unlimitel
> Voxitas
> IP Communications
> Voxbone
> Teliax
>
> Know of others? Love or hate a provider? Hit me either
> privately or on
> list.
>
> Beckman
> ---------------------------------------------------------------------------
> Peter Beckman
> Internet Guy
> beckman@angryox.com
> http://www.angryox.com/
> ---------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by
> http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] US/Canada DID Providers who expose full DID inventory
I'm trying to find more US and Canada (NANPA generally) Wholesale DID
origination providers who, in order of importance to me:
origination providers who, in order of importance to me:
1 expose full DID for inventory
2 immediate provisioning of in-stock DIDs
3 have good quality, reliability and are responsive
4 Have an API
5 offer per-minute pricing, competitive pricing
6 prepaid
Here's the providers I know about already
Vitelity: API, Full DID
Junction: API, NPANXX only
Les.net: No API, Full DID
DIDww: API, NPA only
DIDx: API, Full DID (eww)
Providers I am aware of, but would love comments:
Unlimited Net: API, don't know about exposure
VoicePulse: API, Full DID (I think)
Voice Network: Dunno
Providers that don't expose DID:
Unlimitel
Voxitas
IP Communications
Voxbone
Teliax
Know of others? Love or hate a provider? Hit me either privately or on
list.
Beckman
---------------------------------------------------------------------------
Peter Beckman Internet Guy
beckman@angryox.com http://www.angryox.com/
---------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Asterisk support around the Walnut Creek, CA (Bay Area)
There's a fair number of Asterisk people in the Bay Area, Craigslist
posts work pretty well.
posts work pretty well.
If you can't find someone, I may be able to help, I'm in Emeryville.
On Sep 30, 2008, at 6:55 AM, hin lee wrote:
> Are there any local Asterisk support around the Bay Area? I am
> considering implementing Asterisk PBX in 2-3 months and would love
> to get some assistance.
>
> Thanks!
> Hin
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Eric Chamberlain, Founder
RF.com - http://RF.com/
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Exquisite Replica
King Replica - Best Replica watches, jewellery an accessories!
We specialize in the sales of brand name quality replica timepieces,
at some of the lowest prices possible. With our large selection of watches
you can be sure to find that perfect timepiece that will suit you best.
Re: [asterisk-biz] Asterisk support around the Walnut Creek, CA (Bay Area)
If you cannot find someone local, setting up a system remotely is quote easy as well provided there is someone onsite who can do some basic cabling, install Linux on a box and open SSH. I have done this MANY times and have glowing testimonials for this from small to large systems.
Thanks
Steve Totaro
1.888.777.1888
--
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)
Thanks
Steve Totaro
1.888.777.1888
On Tue, Sep 30, 2008 at 10:26 PM, Rehan Allah Wala <rehan@supertec.com> wrote:
Would you be interested in a hosted asterisk pbx ?
If yes, You can explore ip-pabx.com from my company
It is free for upto 10 extentions, you just pay for usage of mins and phone lines.
Rehan
> Are there any local Asterisk support around the Bay Area? I am considering implementing Asterisk PBX in 2-3 months and would love to get some assistance.>> Thanks!> Hin>>>>> _______________________________________________> --Bandwidth and Colocation Provided by http://www.api-digital.com-->> AstriCon 2008 - September 22 - 25 Phoenix, Arizona> Register Now: http://www.astricon.net>> asterisk-biz mailing list> To UNSUBSCRIBE or update options visit:
Rehan Ahmed AllahWalaMsn/Yahoo/GoogleTalk/Email: Rehan@Rehan.comhttp://www.supertec.com/ - Internet Telephony SolutionsHttp://www.DIDX.net - DID Number Market Place.Don't Remember Me ? Visit http://www.Rehan.com
~~~~~~~~~~~~~~~~~~~"First they ignore you, then they laugh at you, then they fight you, then you win."By Gandhi.
"Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)
[asterisk-biz] Asterisk support around the Walnut Creek, CA (Bay Area)
Are there any local Asterisk support around the Bay Area? I am considering implementing Asterisk PBX in 2-3 months and would love to get some assistance.
Thanks!
Hin
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Monday, September 29, 2008
[asterisk-biz] MOR PRO kolmisoft
Hi,
Does any knows if Kolmisoft still in business I'v been trying to get a hold of them for the past week and no results. When I dial they number is just silent and then gets a busy tone.. no one have responded to my email yet!!!
Do u guys know whats the deal with them?
[asterisk-biz] Routes Available
We've got the following DIRECT routes available at wholesale prices. Please contact us via sales@voicepundit, if you are interested:
| Country |
| Bangladesh |
| Chile Mobile |
| Cuba |
| Djibouti |
| Gambia |
| India |
| Iran |
| Kuwait |
| Myanmar |
| Nepal |
| Oman |
| Pakistan |
| Qatar |
| Saudi Arabia |
| Somalia |
| Sierra Leone |
| Sri Lanka |
| UAE |
[asterisk-biz] FXO Gateway (H.323) Multi-Tech MVP120 $35
Multi-Tech MVP120 1-Port FXO VOIP Gateway (H.323)
I used the multitech for a proof of concept a few years back.
I am willing to sell it for $35 + $5 s&h.
Let me know.
Dave
wallyhts@gmail.com
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Chanel Watches
Why would you want to purchase a replica watch from King-replicas?
There may be many reasons:
1. You want a genuine Rolex / Breitling watch, but the price is too ridiculous
2. You want to impress your friends or business clients
3. You want to keep your original safe, while using the replica for daily wear and tear on it.
Designer replicas and brands available
Buy a replica Rolex at only a fraction of the price!
You can safe 15% if you buy 2 or more watches!
Go to that cocktail party with this watch, and be sure to catch people's attention.
You'll have all the class, and still have all your money.
Franck Muller Watches
Why would you want to purchase a replica watch from King-replicas?
There may be many reasons:
1. You want a genuine Rolex / Breitling watch, but the price is too ridiculous
2. You want to impress your friends or business clients
3. You want to keep your original safe, while using the replica for daily wear and tear on it.
99% brand replicas found here
Replica Watches - cheap and really good solution!
What is a replica watch and how is it different from the real watches?
A replica watch is a watch made similar to that of the real brand ones, except, at a much lower cost.
A real Rolex can go up to hundreds of thousands of dollars, but you can get a replica similar to that one,
for only a few hundred dollars. This allows the normal everyday person to be able to look and feel classy,
without having to actually spend such ridiculous amounts of money on it.
Luxury Pens
King Replica - Best Replica watches, jewellery an accessories!
We specialize in the sales of brand name quality replica timepieces,
at some of the lowest prices possible. With our large selection of watches
you can be sure to find that perfect timepiece that will suit you best.
Sunday, September 28, 2008
[asterisk-biz] Nicaragua DID
Is there anyone out there providing Nicaragua DIDs?
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] India Mobile 2.9 Rest of india 3.2
Hot offer 919 - ALL INDIA MOBILE 2.9 91 - Rest of India 3.2 Contact wimpys_k (at )yahoo.com To Start with Pay as you Go .. Post Pay Facilities will be given after one week. Mexico - Nea & CUBA available Purely on Pre Pay Terms Only(BANK WIRE) -- Sathish C The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above.If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. |
Italian shoes for cheap
Boasting the largest collection of designer shoes ever assembled on the internet, this one stop site has brands ranging from Boss to Gucci to Louis Vuitton.
Designer haute couture that won't leave you penniless, make sure you check this site out for your Christmas shopping.
Your loved ones will adore you for it.
Enter
Re: [asterisk-biz] proper analog behavior using an ATA
Please explain the difference between an anaolg card on a digital System that you have such high regard for? Is it just simply the lack of DSPs on the cards, the quality, what specifically?
They do the same thing in function.
I heard nothing but complaining but not a bit of room to help you. Anyways, selling Channel banks should not be a problem since it sounds like you sell into the pretty high end market.
Thanks,
Steve Totaro
On Sun, Sep 28, 2008 at 5:50 PM, Jim Houser <jhouser@trustamerifirst.com> wrote:
Alex,
Please let your friend know I said "thanks" for his comments. I agree with his thinking.
Jim
----- Original Message -----
From: "Alex Balashov" <abalashov@evaristesys.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz@lists.digium.com>
Sent: Sunday, September 28, 2008 4:42:11 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-biz] proper analog behavior using an ATA
This opinion is offered by a friend of mine who saw the message, but who
is not on the list. Credit goes to Sean McCord (scm!!at!!cycoresys.com).
---
"The _only_ way to do analog right is with dedicated analog
hardware: a channel bank. You can "get it working" with ATAs and PCI
cards, but you will _always_ be fighting problems. There will be plenty
of proponents of ATA and Digium cards for analog saying that they work
just fine. There always are. For a production system in which you will
actually be using features on the phone beyond the basic "connect a call
to the PSTN," and, more importantly, interfacing with other analog gear,
you will only drive yourself crazy and waste more money in time and
effort getting them to work.
Inevitably, the voltages are wrong, the ringing is insufficient, the
transmission levels are incorrect, the signalling isn't reliably passed,
timings are not what they should be, etc. IMHO, these devices are made
by people and companies which are VoIP first and analog
"Johnny-come-lately"'s. I have fought too many battles and have eaten
the purchase of too many channel banks when all else failed to ever
properly recommend ATA for analog PCI cards for most any professional
installation.
This is certainly true of Sipura/Linksys, Grandstream, Digium, and
Sangoma, by my own experience. I am tentatively hopeful of the Rhino
analog PCI cards, but I only have one of those in the field on a system
which has light traffic and limited need of features.
Note that this distrust does not extend to a significant extent to
digital PCI cards. I have generally not had nearly the problems (after
the first or second generation Digium cards, anyway) with them. There
_are_ problems, of course, but they tend to be "hard" problems, not the
intermittant, degrading-over-time types of problems as with the analog
gear, so they can, at least, be dealt with.
Channel bank-based solutions _are_ more expensive on the hardware, and I
don't really see a way around that, but the trade-off is that, over
time, they will be _much_ _less_ expensive due to the lower maintenance
and troubleshooting costs. I know it's a hard sell, but it is
definitely worth it.
As for alternatives to Adits, the CAC AccessBanks (I or II) can be
obtained cheaply on eBay and the like, if you get lucky (though watch
that the ABIs are not TR-08s, which will not work). Rhino also makes
excellent channel banks, and if you are going to buy new equipment (as
opposed to used or eBay eq), they offer a good price, as well."
---
Jim Houser wrote:
> Hi all,
>
> Sorry, kinda long but please read...
>
> I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
>
> I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
>
> Here's my issue I hope to get feedback and help with;
>
> I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
>
> However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
>
> My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
>
> In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times. Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
>
> Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home. That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
>
> I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
>
> What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
>
> Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
>
> Jim
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)
Re: [asterisk-biz] proper analog behavior using an ATA
Steve,
Sorry if my email was packed with TMI. It was not meant to annoy anyone. I just wanted it clear I am not analog stupid and had a solid reference for ATA performance expectations.
> Just for future reference, you don't really compile Asterisk@home, it is just an image that you install.
I said "Started with compiling my own and Asterisk@home." and should have said "started with compiling my own with raw editing of .conf files THEN trying Asterisk@home as a packaged solution".
Thanks for your comments and suggestions.
Jim
----- Original Message -----
From: "Steve Totaro" <stotaro@totarotechnologies.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz@lists.digium.com>
Sent: Saturday, September 27, 2008 2:40:17 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-biz] proper analog behavior using an ATA
Correction, bibliography = biography
On Sat, Sep 27, 2008 at 3:34 PM, Steve Totaro < stotaro@totarotechnologies.com > wrote:
See comments inline.
On Sat, Sep 27, 2008 at 12:42 PM, Jim Houser < jhouser@trustamerifirst.com > wrote:
Hi all,
Sorry, kinda long but please read...
I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
Why the bibliography? I understand that you have been doing telephony for a long time. So have I, but just not as long as you, same understanding though.
I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
Most "Old School Telephony Guys" Hate VoIP. That may not be the case with you, but most people don't like change in general.
Just for future reference, you don't really compile Asterisk@home, it is just an image that you install.
Here's my issue I hope to get feedback and help with;
I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
Give Quintum a try, they are excellent. I have heard good things about Rhino and Xorcom. Do you know they stopped using 25 pair lines for stations, that could be part of the problem ;-P
My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times.
To be fair, it was not plug and play, you had to be somewhat skilled at the switch you were configuring and good with a 110/66 block and a punch too. Many times you needed a couple of people to bust out the toner to make sure you had the right pair.
Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
If I were you and I never will be, but I would try to adapt to the paradigm that VoIP is not behind the times, it is just trying to accommodate other technologies that are behind the times, such as modems and FAXs.
Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home.
My mother can use a computer very well (and she was "trained" on a typewriter) and my niece is amazing with technnology for her age.
Things get old and antiquated. Morse Code, VCRs, reel to reels, switchboards, betamax, even OTA non digitial TV in a few months. Everything changes, gets better, and people cling to the old ways. I know several people that swear that their old records and record player sound better than a digital CD.
That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
And it does, at a MUCH cheaper price point, with many more features, if engineered and installed by someone who knows what they are doing. Flexibility is mind blowing.
I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
I think you should try a few different vendors and solutions. What you mention above is going to give you pretty much the best analog. I am sure someone else can sell you something that does this with SIP. My personal recommendation is Quintum, but I am sure others do just as well.
What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
Generally, you buy a Digium or Sangoma board with the right number of FXO ports and then in install Polycom (or whatever) SIP phones.
Personally, with both Digium and Sangoma FXS ports, I get perfect operation. Maybe try posting your configs instead of a book and biography.
Have you explored your zap configs? There could be a simple setting that makes everything wonderful. Asterisk@home didn't really have much facility for that.
Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
Jim
Thanks,
Steve Totaro
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] proper analog behavior using an ATA
Alex,
Please let your friend know I said "thanks" for his comments. I agree with his thinking.
Jim
----- Original Message -----
From: "Alex Balashov" <abalashov@evaristesys.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz@lists.digium.com>
Sent: Sunday, September 28, 2008 4:42:11 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-biz] proper analog behavior using an ATA
This opinion is offered by a friend of mine who saw the message, but who
is not on the list. Credit goes to Sean McCord (scm!!at!!cycoresys.com).
---
"The _only_ way to do analog right is with dedicated analog
hardware: a channel bank. You can "get it working" with ATAs and PCI
cards, but you will _always_ be fighting problems. There will be plenty
of proponents of ATA and Digium cards for analog saying that they work
just fine. There always are. For a production system in which you will
actually be using features on the phone beyond the basic "connect a call
to the PSTN," and, more importantly, interfacing with other analog gear,
you will only drive yourself crazy and waste more money in time and
effort getting them to work.
Inevitably, the voltages are wrong, the ringing is insufficient, the
transmission levels are incorrect, the signalling isn't reliably passed,
timings are not what they should be, etc. IMHO, these devices are made
by people and companies which are VoIP first and analog
"Johnny-come-lately"'s. I have fought too many battles and have eaten
the purchase of too many channel banks when all else failed to ever
properly recommend ATA for analog PCI cards for most any professional
installation.
This is certainly true of Sipura/Linksys, Grandstream, Digium, and
Sangoma, by my own experience. I am tentatively hopeful of the Rhino
analog PCI cards, but I only have one of those in the field on a system
which has light traffic and limited need of features.
Note that this distrust does not extend to a significant extent to
digital PCI cards. I have generally not had nearly the problems (after
the first or second generation Digium cards, anyway) with them. There
_are_ problems, of course, but they tend to be "hard" problems, not the
intermittant, degrading-over-time types of problems as with the analog
gear, so they can, at least, be dealt with.
Channel bank-based solutions _are_ more expensive on the hardware, and I
don't really see a way around that, but the trade-off is that, over
time, they will be _much_ _less_ expensive due to the lower maintenance
and troubleshooting costs. I know it's a hard sell, but it is
definitely worth it.
As for alternatives to Adits, the CAC AccessBanks (I or II) can be
obtained cheaply on eBay and the like, if you get lucky (though watch
that the ABIs are not TR-08s, which will not work). Rhino also makes
excellent channel banks, and if you are going to buy new equipment (as
opposed to used or eBay eq), they offer a good price, as well."
---
Jim Houser wrote:
> Hi all,
>
> Sorry, kinda long but please read...
>
> I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
>
> I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
>
> Here's my issue I hope to get feedback and help with;
>
> I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
>
> However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
>
> My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
>
> In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times. Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
>
> Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home. That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
>
> I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
>
> What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
>
> Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
>
> Jim
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] proper analog behavior using an ATA
This opinion is offered by a friend of mine who saw the message, but who
is not on the list. Credit goes to Sean McCord (scm!!at!!cycoresys.com).
is not on the list. Credit goes to Sean McCord (scm!!at!!cycoresys.com).
---
"The _only_ way to do analog right is with dedicated analog
hardware: a channel bank. You can "get it working" with ATAs and PCI
cards, but you will _always_ be fighting problems. There will be plenty
of proponents of ATA and Digium cards for analog saying that they work
just fine. There always are. For a production system in which you will
actually be using features on the phone beyond the basic "connect a call
to the PSTN," and, more importantly, interfacing with other analog gear,
you will only drive yourself crazy and waste more money in time and
effort getting them to work.
Inevitably, the voltages are wrong, the ringing is insufficient, the
transmission levels are incorrect, the signalling isn't reliably passed,
timings are not what they should be, etc. IMHO, these devices are made
by people and companies which are VoIP first and analog
"Johnny-come-lately"'s. I have fought too many battles and have eaten
the purchase of too many channel banks when all else failed to ever
properly recommend ATA for analog PCI cards for most any professional
installation.
This is certainly true of Sipura/Linksys, Grandstream, Digium, and
Sangoma, by my own experience. I am tentatively hopeful of the Rhino
analog PCI cards, but I only have one of those in the field on a system
which has light traffic and limited need of features.
Note that this distrust does not extend to a significant extent to
digital PCI cards. I have generally not had nearly the problems (after
the first or second generation Digium cards, anyway) with them. There
_are_ problems, of course, but they tend to be "hard" problems, not the
intermittant, degrading-over-time types of problems as with the analog
gear, so they can, at least, be dealt with.
Channel bank-based solutions _are_ more expensive on the hardware, and I
don't really see a way around that, but the trade-off is that, over
time, they will be _much_ _less_ expensive due to the lower maintenance
and troubleshooting costs. I know it's a hard sell, but it is
definitely worth it.
As for alternatives to Adits, the CAC AccessBanks (I or II) can be
obtained cheaply on eBay and the like, if you get lucky (though watch
that the ABIs are not TR-08s, which will not work). Rhino also makes
excellent channel banks, and if you are going to buy new equipment (as
opposed to used or eBay eq), they offer a good price, as well."
---
Jim Houser wrote:
> Hi all,
>
> Sorry, kinda long but please read...
>
> I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
>
> I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
>
> Here's my issue I hope to get feedback and help with;
>
> I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
>
> However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
>
> My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
>
> In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times. Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
>
> Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home. That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
>
> I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
>
> What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
>
> Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
>
> Jim
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Telephony Depot is hiring
Folks,
Telephony Depot is always looking for talented people who get excited
about open telephony platforms such as Asterisk (in its various
forms), FreeSwitch, Yate etc, and who just plain find telephony
hardware strangely fascinating ;-) Recently though, we've decided to
add more sales, support and programming depth to our bench and have
immediate need for a few people in our Philadelphia office. We're
looking primarily for:
- inside sales rep
- entry-level tech support engineer
- web developer / programmer
but we're a young & growing company (Inc. 500 two years running now)
and we're definitely open to any other strategic hire that might help
take us to the next level of growth. If you think you have something
to contribute to online merchandising of hardware around open
telephony, send your resume and a brief introduction to work@our-
domain-name (sorry email harvesters!). Telecommuting may be an option
for some positions, so if Philadelphia is a non-starter please don't
assume this rules you out!
Sincerely,
--
Darren Nickerson
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax)
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Prestigious shoes for men and women
99% likeness replica shoes can be found at our online stores. No one will be able to tell the difference and you can save lots of money at the same time. Browse for your favorite brand now
Enter
Saturday, September 27, 2008
Italian shoes for cheap
Rock bottom prices with free delivery, these shoes will get you talking .. and buying.
With over 60% of sales from repeat customers who were very satisfied with their initial purchase, the sales of these shoes have caused many designs to go out of stock very quickly.
Don't miss out, get your Christmas shopping done today
Enter
Re: [asterisk-biz] proper analog behavior using an ATA
Correction, bibliography = biography
On Sat, Sep 27, 2008 at 3:34 PM, Steve Totaro <stotaro@totarotechnologies.com> wrote:
See comments inline.On Sat, Sep 27, 2008 at 12:42 PM, Jim Houser <jhouser@trustamerifirst.com> wrote:
Hi all,
Sorry, kinda long but please read...
I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
Why the bibliography? I understand that you have been doing telephony for a long time. So have I, but just not as long as you, same understanding though.
I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
Most "Old School Telephony Guys" Hate VoIP. That may not be the case with you, but most people don't like change in general.
Just for future reference, you don't really compile Asterisk@home, it is just an image that you install.
Here's my issue I hope to get feedback and help with;
I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
Give Quintum a try, they are excellent. I have heard good things about Rhino and Xorcom. Do you know they stopped using 25 pair lines for stations, that could be part of the problem ;-P
My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times.
To be fair, it was not plug and play, you had to be somewhat skilled at the switch you were configuring and good with a 110/66 block and a punch too. Many times you needed a couple of people to bust out the toner to make sure you had the right pair.
Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
If I were you and I never will be, but I would try to adapt to the paradigm that VoIP is not behind the times, it is just trying to accommodate other technologies that are behind the times, such as modems and FAXs.
My mother can use a computer very well (and she was "trained" on a typewriter) and my niece is amazing with technnology for her age.
Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home.
Things get old and antiquated. Morse Code, VCRs, reel to reels, switchboards, betamax, even OTA non digitial TV in a few months. Everything changes, gets better, and people cling to the old ways. I know several people that swear that their old records and record player sound better than a digital CD.And it does, at a MUCH cheaper price point, with many more features, if engineered and installed by someone who knows what they are doing. Flexibility is mind blowing.That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
I think you should try a few different vendors and solutions. What you mention above is going to give you pretty much the best analog. I am sure someone else can sell you something that does this with SIP. My personal recommendation is Quintum, but I am sure others do just as well.
What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
Generally, you buy a Digium or Sangoma board with the right number of FXO ports and then in install Polycom (or whatever) SIP phones.
Personally, with both Digium and Sangoma FXS ports, I get perfect operation. Maybe try posting your configs instead of a book and biography.
Have you explored your zap configs? There could be a simple setting that makes everything wonderful. Asterisk@home didn't really have much facility for that.
Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
Jim
Thanks,
Steve Totaro
Subscribe to:
Posts (Atom)
