Monday, March 31, 2008

[asterisk-biz] Solving DTMF issue

I would like to hire an expert who can absolutely fix the following issue:

I have SIP providers who sends several DTMF mode to my asterisk at
different times. We have changed "dtmfmode=auto" and still sometimes
it recognized dtmf and sometimes not. I need someone to identify the
issue and detect all DTMF mode sent by the SIP provider no matter what
it is and pass it to asterisk without any problem.

Thanks

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Re: [asterisk-biz] Need White Label Hosted PBX!

is 25% respectable ?

try ip-pabx.com I can try to get you re seller ship for it, it is one of supertec.com product.

Rehan



> Hello,
>
> Can any one recommend a white label hosted PBX service where the per
> extension and minute margins are respectable for resellers. If so please
> let me know. If you offer such a service especially in EU please contact
> me off list. This is fairly urgent.
>
> Best Regards,
>
> Robert
>
>
> _______________________________________________
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>
> 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.
 

Re: [asterisk-biz] Simulating 911 ANI/ALI

Actually I am biding for the project and I am in between the provider and the customer. The customer wants me to do a demonstration first as a proof of concept but the data will be subject to the final confirmation by the provider. Until then I won't be able to talk to the provider directly as it is masked by the customer. Any suggestions?

asterisk_help@iwishi.nu wrote:
... I plan to use Asterisk as the front end to  connect to a provider who will connect via SIP trunk and pass all 911 calling  informations like... 1. ANI (Automatic Numbering Information) 2. ALI (Automatic Location Information) a. Caller no b. Building name / caller name c. Address d. Latitude and Longitude of the caller address  3. Incident Information a. Incident code b. Incident Description. c. might have other information as well.  Then I wish to pass these through the manager interface where it can be  collected and processed into a database server to display it on a console...  perhaps like a crm pop-up.     
  You will need to contact the provider that will send these details via SIP  and ask of the standard they will be following.  I'm not aware of any  single standard that will address the information you are expecting.  You might want to review:  http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands#SIPcommands  http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header Synopsis - Gets the specified SIP header  http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPGetHeader With this app, you can pick any header from an incoming invite and stuff it into a channel variable. It is a generic way of supporting any  header a vendor or service provider may add that you want to use in your  dialplan.  In the US, the PSAP (Public Safety Answering Provider/Point) is given the  ANI (an identification number, normally a billing phone number) with the  telephone call and they must then use a seperate communications circuit connecting them to a database provider to query for the information needed  to dispatch the call.  Please let me know what standard or spec they are using in their SIP  calls. As a CLEC and VoIP service provider myself, I'm always interested  in learning of new developments in this area.     -Eric Osterberg     Sound Choice Communications LLC      Minnesota, US  _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com--  asterisk-biz mailing list To UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-biz    

Re: [asterisk-biz] activex softphone vs. 1ezphone

Don't you mean OUR new skin?

as in "Or email me and _I_ will setup profile for to make a few calls"

regards,

Drew



Bob G wrote:
I LIKE ONE 1EZPHONE BETTER.
 
YOU SHOULD SEE ALL FEATURES ON THEIR NEW SKIN !
 
 


 
----- Original Message -----
From: "Trixter aka Bret McDanel"
To: "Commercial and Business-Oriented Asterisk Discussion"
Subject: Re: [asterisk-biz] activex softphone
Date: Thu, 27 Mar 2008 18:35:39 +0100



On Thu, 2008-03-27 at 17:13 +0000, Tim Panton wrote:
> On 21 Mar 2008, at 10:38, Andor Czafik (Akakiko) wrote:
>
> > Hi!
> >
> > I need control (answer, and call) sip phone from web browser, and the
> > best is, (i think) the activex softphone.
> > What is the best, and cheaper(or free) activex softphone?
> > Thanks
> > Andor
> >
> > ________________
>
> Does it have to be SIP ? Would IAX do ?
> Do you care about the other 25% of users who use Firefox or Macs?
> Will your users agree to install an activeX control ?


firefox in windows can do activex with a plugin. This also works with
wine, and I think transgaming the makers of cedega (based on the
original MIT licensed wine) are the ones responsible for it.

The final one is probably the biggest, if its not signed its unlikely to
be trusted by many, so there is an additional fee for that. Even if it
is signed many may not want to do that or their corporate security
policy restricts it.

--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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--  Drew Gibson  Systems Administrator OANDA Corporation www.oanda.com 

Re: [asterisk-biz] SIP DDI in Serbia , Moldova, Ukraine, Turkey & BieloRussia

I think didx.net has ukraine and also didww.com


>
>
>
> hi All,
> i am looking for DID in the following countries:
>
> 1. Serbia
> 2. Moldova
> 3. Ukraine
> 4. BieloRussia
> 5. Turkey
>
> anyone that can offer these on a high quality , please contact me offlist.
>
> Many thanks,
>
> Mickey



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.
 

Re: [asterisk-biz] activex softphone vs. 1ezphone

On Mon, 2008-03-31 at 15:12 -0500, Bob G wrote:
> I LIKE ONE 1EZPHONE BETTER.
>
> YOU SHOULD SEE ALL FEATURES ON THEIR NEW SKIN !
>
> http://demo.arkinfotec.com/projects/skin_design/index.html
>

OMGWTFROTFLMAO!@%$@!%!@%!#@^!#^

you do realize that all caps like you did is considered yelling by most
people on the net right? And that yelling such as that generally makes
business types less than eager to follow the link to see new skin
features (although it has the opposite effect if its a skin flick :)

--
Trixter http://www.0xdecafbad.com

Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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Re: [asterisk-biz] activex softphone vs. 1ezphone

I LIKE ONE 1EZPHONE BETTER.
 
YOU SHOULD SEE ALL FEATURES ON THEIR NEW SKIN !
 
 


 
----- Original Message -----
From: "Trixter aka Bret McDanel"
To: "Commercial and Business-Oriented Asterisk Discussion"
Subject: Re: [asterisk-biz] activex softphone
Date: Thu, 27 Mar 2008 18:35:39 +0100



On Thu, 2008-03-27 at 17:13 +0000, Tim Panton wrote:
> On 21 Mar 2008, at 10:38, Andor Czafik (Akakiko) wrote:
>
> > Hi!
> >
> > I need control (answer, and call) sip phone from web browser, and the
> > best is, (i think) the activex softphone.
> > What is the best, and cheaper(or free) activex softphone?
> > Thanks
> > Andor
> >
> > ________________
>
> Does it have to be SIP ? Would IAX do ?
> Do you care about the other 25% of users who use Firefox or Macs?
> Will your users agree to install an activeX control ?


firefox in windows can do activex with a plugin. This also works with
wine, and I think transgaming the makers of cedega (based on the
original MIT licensed wine) are the ones responsible for it.

The final one is probably the biggest, if its not signed its unlikely to
be trusted by many, so there is an additional fee for that. Even if it
is signed many may not want to do that or their corporate security
policy restricts it.

--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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Re: [asterisk-biz] Need White Label Hosted PBX!

Rehan,

25% margin is respectable but what is it for? Outgoing calls, extensions? I can't find back end admin demo.

regards,

robert

Rehan Allah Wala wrote:
is 25% respectable ?

try ip-pabx.com I can try to get you re seller ship for it, it is one of supertec.com product.

Rehan



> Hello,
>
> Can any one recommend a white label hosted PBX service where the per
> extension and minute margins are respectable for resellers. If so please
> let me know. If you offer such a service especially in EU please contact
> me off list. This is fairly urgent.
>
> Best Regards,
>
> Robert
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:



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.
 
 
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[asterisk-biz] Need White Label Hosted PBX!

Hello,

Can any one recommend a white label hosted PBX service where the per
extension and minute margins are respectable for resellers. If so please
let me know. If you offer such a service especially in EU please contact
me off list. This is fairly urgent.

Best Regards,

Robert


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Re: [asterisk-biz] Simulating 911 ANI/ALI

> ... I plan to use Asterisk as the front end to
> connect to a provider who will connect via SIP trunk and pass all 911 calling
> informations like...
> 1. ANI (Automatic Numbering Information)
> 2. ALI (Automatic Location Information)
> a. Caller no
> b. Building name / caller name
> c. Address
> d. Latitude and Longitude of the caller address
>
> 3. Incident Information
> a. Incident code
> b. Incident Description.
> c. might have other information as well.
>
> Then I wish to pass these through the manager interface where it can be
> collected and processed into a database server to display it on a console...
> perhaps like a crm pop-up.


You will need to contact the provider that will send these details via SIP
and ask of the standard they will be following. I'm not aware of any
single standard that will address the information you are expecting.

You might want to review:

http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands#SIPcommands

http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
Synopsis - Gets the specified SIP header

http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPGetHeader
With this app, you can pick any header from an incoming invite and
stuff it into a channel variable. It is a generic way of supporting any
header a vendor or service provider may add that you want to use in your
dialplan.

In the US, the PSAP (Public Safety Answering Provider/Point) is given the
ANI (an identification number, normally a billing phone number) with the
telephone call and they must then use a seperate communications circuit
connecting them to a database provider to query for the information needed
to dispatch the call.

Please let me know what standard or spec they are using in their SIP
calls. As a CLEC and VoIP service provider myself, I'm always interested
in learning of new developments in this area.


-Eric Osterberg
Sound Choice Communications LLC
Minnesota, US

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Re: [asterisk-biz] Simulating 911 ANI/ALI

Thanks for your response.

I have a project... that why. I plan to use Asterisk as the front end to connect to a provider who will connect via SIP trunk and pass all 911 calling informations like...
1. ANI (Automatic Numbering Information)
2. ALI (Automatic Location Information)
 a. Caller no
 b. Building name / caller name
 c. Address
 d. Latitude and Longitude of the caller address

3. Incident Information
 a. Incident code
 b. Incident Description.
 c. might have other information as well.

Then I wish to pass these through the manager interface where it can be collected and processed into a database server to display it on a console... perhaps like a crm pop-up.

Since I have to prove that asterisk can do the job, I need to show that it will work by simulating it. Any suggestions?




asterisk_help@iwishi.nu wrote:

On Sat, 29 Mar 2008, Si Tai Fan wrote:
Can anyone suggest how I can simulate a 911 call with a ALI (Automatic Numbering Information) over SIP to Asterisk? Will Asterisk be able to receive this information?

OK, first... I think of asterisk as "the telephoney toolkit" rather than a PBX.  The answer to "can asterisk do that" is always yes. But somethings are easier to do. Recall you have source code so you can do anything.

ALI != Automatic NUMBERING Information

What you are asking about is huge!  I think standards are still emerging so you really need to define exactly what you want.

Try reviewing these publications first: http://www.its.dot.gov/ng911/ng911_pubs.htm

Esp look over: NG9-1-1 System Description and Requirements Document

Then use google and the following terms: "LoST sip 911 i3 ecrit"

 -Eric Osterberg
 -Sound Choice Communications LLC
   Minnesota, US
 
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Re: [asterisk-biz] Looking for A-Z (SIP)

Hi Cliff,

You may also want to checkout www.voipinvite.com

We offer Least Cost rates using multiple Tier 1s and our own foot print for international wholesale sip termination.

Billing increments are 1/1 with 24x7 support and live CDR/CRM portal. Please fill out our inquiry form on the website for free test credit and pricing.

Thank you,

Vijay



-----Original Message-----
From: Cliff Mazer <cmazer@thinkingphones.com>
Sent: Monday, March 31, 2008 9:46am
To: Commercial and Business-Oriented Asterisk Discussion <asterisk-biz@lists.digium.com>
Subject: Re: [asterisk-biz] Looking for A-Z (SIP)

Moshe,
Can you tell me a little about how you are doing QOS.

Cliff

-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com
[mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Moshe Maeir
Sent: Monday, March 31, 2008 3:46 AM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] Looking for A-Z (SIP)

You may want to get a free test account from us at
www.flatplanetphone.com.
I think our rates are competitive.
Moshe

Dome Charoenyost wrote:
> Dear All,
> I'm looking for A-Z provider. but price must be close RNK
> (rnktel.com). If someone can do please contact me directly.
>
> Best Regards.
>
> Dome C.
>
> _______________________________________________
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>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>

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Re: [asterisk-biz] Looking for A-Z (SIP)

Moshe,
Can you tell me a little about how you are doing QOS.

Cliff

-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com
[mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Moshe Maeir
Sent: Monday, March 31, 2008 3:46 AM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] Looking for A-Z (SIP)

You may want to get a free test account from us at
www.flatplanetphone.com.
I think our rates are competitive.
Moshe

Dome Charoenyost wrote:
> Dear All,
> I'm looking for A-Z provider. but price must be close RNK
> (rnktel.com). If someone can do please contact me directly.
>
> Best Regards.
>
> Dome C.
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>

http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>

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Re: [asterisk-biz] Looking for A-Z (SIP)

You may want to get a free test account from us at www.flatplanetphone.com.
I think our rates are competitive.
Moshe

Dome Charoenyost wrote:
> Dear All,
> I'm looking for A-Z provider. but price must be close RNK
> (rnktel.com). If someone can do please contact me directly.
>
> Best Regards.
>
> Dome C.
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>

http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>

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Sunday, March 30, 2008

Re: [asterisk-biz] IP / USB Phones required

Cliff,
If you convert 25% of your calls to sales you are great!
Moshe

Cliff Mazer wrote:

Hi Andy,

 

I know how you feel as I been in sales for over 30yrs. I can only tell you that I work with a funnel. Only 25% of the people who show an interest will buy. That’s why I look for business every day. I call 70 people a day or more to get a enough in my funnel to make sure If a  deal doen't make it to the end, I have enough to cover it. Selling is not easy.

 

Cliff

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Andy Spring
Sent: Sunday, March 23, 2008 9:47 AM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] IP / USB Phones required

 


Hi All,

 

I am depressed these days, and I am always worried about my business and can not sleep well and eating well, I am always thinking of some questions, after I have posted our products here, many many people enquiry and I am also very happy and patient to give them answers but they never give orders, I do not know why this situation happens and what our customers are worring about or thinking about,

can anyone give me an answer?

 

Thanks and best regards

andy

 
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Saturday, March 29, 2008

Re: [asterisk-biz] MagicJack

I read somewhere that they have CLEC status in most of the states, so they maybe able to terminate calls for almost nothing.
But even if termination cost you nothing, it is still very hard to make money on $20 a year. Even if your costs are zero...

Nitzan Kon wrote:
Matthew Rubenstein wrote: 	Anyone know any more details? How do they offer $20:year, when most VoIP competitors charge at least $15-35:month? Are they using Asterisk for infrastructure - any thing more than maybe just     
voicemail?  It's $20/yr, but you have to buy their device ($40) that connects to your computer via USB. They then (plan to) proceed to spam you with ads using the software you need to install to run the device. You cannot use an ATA or any SIP device with their service as far as I know.  They do have CLEC status in some state (all of them?), look up YMAX Corp. So they should be getting some price breaks (and reciprocal revenue?) from that. But I think their main business model is to steal as much customers and start spamming them with ads.  I dunno about this business model though. They're basically selling service way below cost, assuming they can recoup it back in advertisements. I doubt this will work long term.  More likely, they'll go out of business, taking all the cash they amassed with them and leaving the users to hang out dry. I've seen some people buy ahead 5 years of service with their device. Dunno about everyone else but I'm not so sure they'll be able to stick for 5 MONTHS.. let alone 5 years.  (having said that, they started last year, and are backed by a guy with pretty deep pockets. So who knows...)  -- Nitzan Kon, CEO Future Nine Corporation www.future-nine.com   _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com--  asterisk-biz mailing list To UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-biz    

Re: [asterisk-biz] Simulating 911 ANI/ALI

On Sat, 29 Mar 2008, Si Tai Fan wrote:
> Can anyone suggest how I can simulate a 911 call with a ALI (Automatic
> Numbering Information) over SIP to Asterisk? Will Asterisk be able to receive
> this information?

OK, first... I think of asterisk as "the telephoney toolkit" rather than a
PBX. The answer to "can asterisk do that" is always yes. But somethings
are easier to do. Recall you have source code so you can do anything.

ALI != Automatic NUMBERING Information

What you are asking about is huge! I think standards are still emerging
so you really need to define exactly what you want.

Try reviewing these publications first:

http://www.its.dot.gov/ng911/ng911_pubs.htm

Esp look over: NG9-1-1 System Description and Requirements Document

Then use google and the following terms: "LoST sip 911 i3 ecrit"

-Eric Osterberg
-Sound Choice Communications LLC
Minnesota, US

Friday, March 28, 2008

[asterisk-biz] Simulating 911 ANI/ALI

Hi

Can anyone suggest how I can simulate a 911 call with a ALI (Automatic Numbering Information) over SIP to Asterisk? Will Asterisk be able to receive this information?

Thanks in advance...
Si

Re: [asterisk-biz] ITSP Billing Increments....?

Let's put it in even simpler terms. True story by the way: Spain cellular
sells for 19cents per minute the cheapest anyone can find (this is only an
example)... So I calculate I can resell for 23cents per minute. But then
everybody tells me that they prefer to buy from X, who was selling at 18
cents. It took me time to figure it out, but now I know: I billed at 6
cents increments and they billed at 60 seconds increments, and probably
their minutes lasted 55 seconds. It worked for them beautifully since
average length of a call on that route was 6 minutes.

Conclusion: the cheapest quality routes charge 30/30 or 60/60 to cover for
the cheaper price you think you are getting.

C. Savinovich


-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com
[mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Linus Surguy
Sent: Friday, March 28, 2008 3:00 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] ITSP Billing Increments....?

> We've been using carriers that bill 1/1, and the quality hasn't been too
> good, lots of 503's etc. We just started testing with someone else that
> offers much better ASR's (they failover internally between qwest, global
> crossing etc), but they bill 30/6. According to THEM, only companies that
> can't offer top quality routes bill 1/1.

Well, I can't speak for the US, but as far as Europe is concerned that is
complete nonsense. All the top tier operators in Europe bill per second.

Linus


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Re: [asterisk-biz] ITSP Billing Increments....?

On Fri, 2008-03-28 at 18:32 -0700, C. Savinovich wrote:
> Let's put it in even simpler terms. True story by the way: Spain cellular
> sells for 19cents per minute the cheapest anyone can find (this is only an
> example)... So I calculate I can resell for 23cents per minute. But then
> everybody tells me that they prefer to buy from X, who was selling at 18
> cents. It took me time to figure it out, but now I know: I billed at 6
> cents increments and they billed at 60 seconds increments, and probably
> their minutes lasted 55 seconds. It worked for them beautifully since
> average length of a call on that route was 6 minutes.
>
> Conclusion: the cheapest quality routes charge 30/30 or 60/60 to cover for
> the cheaper price you think you are getting.


I would put that in the same category as all tier 1 dont do 6/6 or 1/1.
It may be true for some, but I dont think that its true for all (nor do
I believe the reverse of that). It depends on various factors, how you
interconnect, where in some situations, what plan you can convince the
sales rep to give you, etc. The same carrier may do 1/1 and 6/6 but
charge different rates for each and the rates may be so different that
1/1 isnt worth it.

You also said that 19 cents is the cheapest anyone can find, but that
someone is offering it for 18 cents, so by finding the provider that
offers it for 18 cents wouldnt that mean that you found something
cheaper than 19? Ok tangent safe to ignore this paragraph :)

If you look at it this way, statistically speaking 50% of the calls will
be in the lower 50% of the bill increment and 50% of the calls will be
in the higher 50% of the bill increment. Yes statistics dont always
match the real world, but it works as a base guage, and generally they
should match up with a large enough sample.


Assumptions made: the carrier is billing in 1/1 increments. The itsp
is billing 60/60. Cost is 19 cents, its sold at 18 cents.

50% average slack time per final minute means that 9 cents of slack
money is generated.

With a 6 minute average the charge to the customer would be 0.18*6=1.08.
The charge to the ITSP would be (5.5 minutes given the 50% statistical
average listed above) 5.5*0.19=1.045, for a grand profit off that one
call of 3.5 cents.

If you had 20 minute averages you would see a loss of 10.5 cents. For
each 20 minute call you need three 6 minute ones to make up for it. The
break even point with these numbers would be about 9.5 minutes, above
that you lose money on the call, below that you get a minor profit.

Yeah do a few thousand calls per day at the 6 minute average and you
might be able to pay your bills, 5000 calls is only 175
[dollars/euro/etc] per day, which really isnt enough for rack space,
bandwidth, office space, support staff, etc. You would be lucky to
break 5000/mo off that.

Now if you are big enough and doing enough total volume to enough
different places and the math is similar you may be able to make a
reasonable profit, making up in volume what you dont get per call.

Smaller operators would have a problem just paying their bills let alone
salaries however. Larger operators can generally negotiate better rates
as well, which in turn makes them more profitable however they generally
have more overhead which can reverse this trend.

You also have a huge exposure, if the 6 minute trend goes up, even by a
minute, you will see a huge downturn in your income (about 16%), and you
are really close to the border where you lose money on each call.

In short you dont make enough to sell below cost by doing that unless
you only get short calls or your initial increment is much longer than
the average call (like telecom usa with their 20 minute first increment,
and the average voice call when they advertised heavily was 4.3
minutes).

Where slack billing should come in is for gravy. If they charged 19
cents to beat your 23, they would never lose money (on the call itself),
and would make a small amount per call, in this example it would be
about an average of 9.5 cents per call, far better than the 3.5 cents
above. Do a few thousand calls per day and the gravy pot starts to fill
up. It is likely that they have a deal somehow somewhere for a cost
closer to 18 cents, perhaps with a direct connect to the carriers,
perhaps something else.

Now if you offer a "regular" and a "premium" service level where the
only real difference is the bill increment and priority (ie regular can
be dumped in lieu of a premium call, or just wont go through if there
arent enough channels available, etc) you can bill regular at carrier
cost using slack to make some extra, where premium is billed at a higher
rate, so the profit comes from the additional amount charged.

In this way you get customers who are willing to pay more for better
quality, as well as the ones that arent willing to pay more, and instead
of having gobs of idle time on your system you turn a slight profit.

Now if you also had a lower level than regular you could sell idle
capacity potentially below cost (total cost, bandwidth, servers, support
staffers, as well as termination charges) to lose $500 is better than to
lose $1000 so getting something in *some* situations can be better than
getting nothing. When it comes to phone minutes (and network bandwidth,
and ...) if you dont use it at this moment in time its gone forever, you
can never recapture minutes that have already passed.

The rapidly falling costs of bandwidth, rack space, etc is making the
last option less desirable because the costs involved to actually
operate have dropped. They will never reach zero, but in time they will
approach zero.

It used to be that it took millions to get even a small infrastructure,
and operating costs per day were quite high with all the skilled labour
that was required. Those carriers are more likely to give "best effort"
quality just to slow the attrition of funds from their bank accounts.
New providers, especially ITSPs, are far less likely to see any value in
that.

--
Trixter http://www.0xdecafbad.com

Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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Re: [asterisk-biz] ITSP Billing Increments....?

On Fri, 2008-03-28 at 22:00 +0000, Linus Surguy wrote:
> > We've been using carriers that bill 1/1, and the quality hasn't been too
> > good, lots of 503's etc. We just started testing with someone else that
> > offers much better ASR's (they failover internally between qwest, global
> > crossing etc), but they bill 30/6. According to THEM, only companies that
> > can't offer top quality routes bill 1/1.
>
> Well, I can't speak for the US, but as far as Europe is concerned that is
> complete nonsense. All the top tier operators in Europe bill per second.

Yeah this is largely a US conversation based on the initial poster,
although its not bad to comment on how carriers in other parts of the
world do it for comparison. Back in the 80s many carriers refused to do
business service in the US with less than 6/6, I think MCI was the first
to offer that to compete with AT&T who was doing 60/60 at the time.
They claimed at the time they couldnt do less because their billing
systems wouldnt let them, the custom written billing systems that they
have ...

--
Trixter http://www.0xdecafbad.com

Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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Re: [asterisk-biz] ITSP Billing Increments....?

> We've been using carriers that bill 1/1, and the quality hasn't been too
> good, lots of 503's etc. We just started testing with someone else that
> offers much better ASR's (they failover internally between qwest, global
> crossing etc), but they bill 30/6. According to THEM, only companies that
> can't offer top quality routes bill 1/1.

Well, I can't speak for the US, but as far as Europe is concerned that is
complete nonsense. All the top tier operators in Europe bill per second.

Linus


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Re: [asterisk-biz] ITSP Billing Increments....?

In my experience:
 
30/6 is always Tier 1 TDM
6/6 USA
60/60 Mexico
 
1/1 has always been grey routes that are great one day, and terrible the next. Damn Lucent and their TNT's :)
 
-Jon
 
 
----- Original Message -----
Sent: Friday, March 28, 2008 2:00 PM
Subject: Re: [asterisk-biz] ITSP Billing Increments....?


>Hi Doug,
>
>Quoting Douglas Garstang <dougmig33@yahoo.com>:
>
>> So, I have a general question.
>>
>> What billing increments do ITSP's who terminate SIP->PSTN normally bill in?
>>
>> We've been using carriers that bill 1/1, and the quality hasn't been
>> too good, lots of 503's etc. We just started testing with someone
>> else that offers much better ASR's (they failover internally between
>> qwest, global crossing etc), but they bill 30/6. According to THEM,
>> only companies that can't offer top quality routes bill 1/1.
>>
>> What's the deal here? Is this true?
>
>Well unless you are talking to TIER 1 Carriers (AT&T, Verizon, Level3,
>MCI) you are talking to middle men, who may promise 1/1 but would give
>a real hard time on ASRs.

We've been using Verizon to terminate to Russia. Their ASR's have been HORRIBLE. They would regularly reject 50% or more of calls during peak times with SIP server side final responses (500+). They informed us they don't offer any sort of SLA's... pretty funny for a tier one carrier. How can you report problems when there is no standard by which to measure quality? Qwest offer SLA's, but it's so damn complicated, we are still trying to understand it.

Doug.




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Re: [asterisk-biz] ITSP Billing Increments....?


>Hi Doug,
>
>Quoting Douglas Garstang <dougmig33@yahoo.com>:
>
>> So, I have a general question.
>>
>> What billing increments do ITSP's who terminate SIP->PSTN normally bill in?
>>
>> We've been using carriers that bill 1/1, and the quality hasn't been
>> too good, lots of 503's etc. We just started testing with someone
>> else that offers much better ASR's (they failover internally between
>> qwest, global crossing etc), but they bill 30/6. According to THEM,
>> only companies that can't offer top quality routes bill 1/1.
>>
>> What's the deal here? Is this true?
>
>Well unless you are talking to TIER 1 Carriers (AT&T, Verizon, Level3,
>MCI) you are talking to middle men, who may promise 1/1 but would give
>a real hard time on ASRs.

We've been using Verizon to terminate to Russia. Their ASR's have been HORRIBLE. They would regularly reject 50% or more of calls during peak times with SIP server side final responses (500+). They informed us they don't offer any sort of SLA's... pretty funny for a tier one carrier. How can you report problems when there is no standard by which to measure quality? Qwest offer SLA's, but it's so damn complicated, we are still trying to understand it.

Doug.




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Re: [asterisk-biz] ITSP Billing Increments....?

>On Thu, 2008-03-27 at 22:56 -0700, Douglas Garstang wrote:
>> So, I have a general question.
>>
>> What billing increments do ITSP's who terminate SIP->PSTN normally
>> bill in?
>>
>> We've been using carriers that bill 1/1, and the quality hasn't been
>> too good, lots of 503's etc. We just started testing with someone else
>> that offers much better ASR's (they failover internally between qwest,
>> global crossing etc), but they bill 30/6. According to THEM, only
>> companies that can't offer top quality routes bill 1/1.
>>
>> What's the deal here? Is this true?
>
>First, does it really matter?  How many calls do you make that are less
>than 30 seconds? Will the 6 second increment really affect your total
>price?  If the new price is higher, is the quality worth it?

I don't know why, but yes, a lot of our calls are short. Just about all our volume is international, and I guess since a large number of our users are on cell phones, that may somehow be a factor.

>
>
>Since few PSTN based carriers offer 1/1 billing (I think this is
>changing though) on most plans it may be true that you cant do 1/1 with
>quality carriers as a general rule.  I do not however believe it to be
>an absolute rule, and again there is cost vs quality.

Teleglobe and Arbinet do. I think Verizon does as well.




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Re: [asterisk-biz] ITSP Billing Increments....?

There are Tier 1 USA carriers that will terminate 1/1 on an ICB basis. Many
of the resellers of USA Tier 1's will bill USA termination in 1/1 on special
request, especially if you're buying international from them. And of course,
if you have a volume of minutes. For those who claim that when you are
offered 1/1 billing for USA or international termination that you are
getting a substandard route, well keep on buying in 30/6!

Regards,

Jay Kordic
The Horizon Group
Wholesale VOIP/TDM routes/Wholesale IP Bandwidth
1-951-744-9220
1-515-322-0273(fax)
MSN IM- Jaykordic@hotmail.com

March Madness specials
1.ALASKA TERMINATION as low as .0105minute with 90%+ASR's.

2.USA termination offered with a virtual local ANI/CLI.The rates as low as
.004/minute(10 million minute minimum per month for this rate)with 1/1
billing increments.Other rates are available (.0045/.005/.0055) for a lower
commitment.This route covers 115,000 npa/nxx's.The underlying carrier on
this route is a USA Tier 1 provider.

3.USA termination offered with a virtual local ANI/CLI.This route covers
48,732 npa/nxx's in 72 markets and has a rate of .003/minute with 1/1
billing increments.The underlying provider is a USA Tier 1 provider.

4.Internet bandwidth from USA Tier 1 providers for Fast E/Gig E as low as
$14/mb.Includes XO,GX,L3,Qwest,UUNet and more.

5.8XX termination access compensation(.002 -.006/minute)/CABS
compensation(.002-.035/minute).

NEED COLOCATION ? Ask me about it.
Atlanta, GA
Charlotte, NC
Chicago, IL - 350 E. Cermak Road
Chicago, IL - 600 S. Federal Street
Dallas, TX - 2323 Bryan Street
Dallas, TX - 8435 Stemmons Fwy
Los Angeles, CA
Miami, FL
NYC - 60 Hudson Street
NYC - 111 Eighth Avenue
Phoenix, AZ
San Francisco, CA
Santa Clara, CA
Weehawken, NJ

-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com
[mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Mitul Limbani
Sent: Friday, March 28, 2008 2:02 AM
To: asterisk-biz@lists.digium.com
Subject: Re: [asterisk-biz] ITSP Billing Increments....?

Hi Doug,

Quoting Douglas Garstang <dougmig33@yahoo.com>:

> So, I have a general question.
>
> What billing increments do ITSP's who terminate SIP->PSTN normally bill
in?
>
> We've been using carriers that bill 1/1, and the quality hasn't been
> too good, lots of 503's etc. We just started testing with someone
> else that offers much better ASR's (they failover internally between
> qwest, global crossing etc), but they bill 30/6. According to THEM,
> only companies that can't offer top quality routes bill 1/1.
>
> What's the deal here? Is this true?

Well unless you are talking to TIER 1 Carriers (AT&T, Verizon, Level3,
MCI) you are talking to middle men, who may promise 1/1 but would give
a real hard time on ASRs.

Also as you may see, its all mathematics 30/6 and they can route you
through different carriers (mebbe sometimes even on premium routes, coz
their math allow them to do so.)

Again, the good ol saying is still true - "You get wat you pay for !!"

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions,
The Enterprise Linux Company (TM),
www.enterux.com
(P.S.: These are my views)

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[asterisk-biz] 911 Simulating ANI/ALI

Hi

Can anyone suggest how I can simulate a 911 call with a ALI (Automatic Numbering Information) over SIP to Asterisk? Will Asterisk be able to receive this information?

Thanks in advance...
Si

Re: [asterisk-biz] ITSP Billing Increments....?

Hi Doug,

Quoting Douglas Garstang <dougmig33@yahoo.com>:

> So, I have a general question.
>
> What billing increments do ITSP's who terminate SIP->PSTN normally bill in?
>
> We've been using carriers that bill 1/1, and the quality hasn't been
> too good, lots of 503's etc. We just started testing with someone
> else that offers much better ASR's (they failover internally between
> qwest, global crossing etc), but they bill 30/6. According to THEM,
> only companies that can't offer top quality routes bill 1/1.
>
> What's the deal here? Is this true?

Well unless you are talking to TIER 1 Carriers (AT&T, Verizon, Level3,
MCI) you are talking to middle men, who may promise 1/1 but would give
a real hard time on ASRs.

Also as you may see, its all mathematics 30/6 and they can route you
through different carriers (mebbe sometimes even on premium routes, coz
their math allow them to do so.)

Again, the good ol saying is still true - "You get wat you pay for !!"

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions,
The Enterprise Linux Company (TM),
www.enterux.com
(P.S.: These are my views)

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[asterisk-biz] Looking for A-Z (SIP)

Dear All,
I'm looking for A-Z provider. but price must be close RNK
(rnktel.com). If someone can do please contact me directly.

Best Regards.

Dome C.

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Re: [asterisk-biz] ITSP Billing Increments....?

hi,
first of all, as far as i know, there are a lot of short calls. think
of all the calls that end in a voicemail box where the caller doesn't
leave a message. There is a lot of slack in 30 second minimums.

and, historically, 6/6 was used because 6 seconds is 10% of a minute -
it just makes the billing much easier - move the decimal point over 1
digit and that's it. 1/60 = 0.0166 (bar) which makes for one fugly
increment and means everything has to be rounded to reach a number
that doesn't have an infinite fraction at the end of it.

-yair

On Fri, Mar 28, 2008 at 9:52 AM, Trixter aka Bret McDanel
<trixter@0xdecafbad.com> wrote:
>
> On Thu, 2008-03-27 at 22:56 -0700, Douglas Garstang wrote:
> > So, I have a general question.
> >
> > What billing increments do ITSP's who terminate SIP->PSTN normally
> > bill in?
> >
> > We've been using carriers that bill 1/1, and the quality hasn't been
> > too good, lots of 503's etc. We just started testing with someone else
> > that offers much better ASR's (they failover internally between qwest,
> > global crossing etc), but they bill 30/6. According to THEM, only
> > companies that can't offer top quality routes bill 1/1.
> >
> > What's the deal here? Is this true?
>
> First, does it really matter? How many calls do you make that are less
> than 30 seconds? Will the 6 second increment really affect your total
> price? If the new price is higher, is the quality worth it?
>
>
> I am unsure if there is a relation between billing increments and
> quality, but ok.
>
> As for what a provider bills at that is up to the provider itself. This
> may be passed on from the carriers they are using to terminate calls.
> Generally on the PSTN (at least in the US) you will see a higher first
> minute and lower additional minutes. This is because to set up the call
> a few queries have to be made which cost the carriers money. Even if
> the phone isnt answered there are these costs. So what they do is
> charge a higher first minute to recover these costs and when the phone
> is never answered they write that off into other costs which is paid for
> in part by that first minute and each additional minute.
>
> Now most residential service in the US is billed in whole minute
> increments while many business plans are 6 second increments. Really
> there isnt any reason for them to not be each second increments.
>
> For short duration calls a higher billing increment is more profitable,
> which means that other costs can be absorbed by that.
>
> Since few PSTN based carriers offer 1/1 billing (I think this is
> changing though) on most plans it may be true that you cant do 1/1 with
> quality carriers as a general rule. I do not however believe it to be
> an absolute rule, and again there is cost vs quality.
>
> Its very likely that their carriers are doing 30/6 or 6/6 to them, which
> are (or at least were) very common business billing plans. I dont know
> why they did 6 seconds, the carriers claimed it had something to do with
> the way that duration was calculated over a decade ago, that the systems
> would not let them do 1/1. I didnt believe it then, and certainly now
> its not true, or if it is there is no real reason for it to be true
> (intercarrier settlements are often enough millisecond resolution
> showing that it doesnt have to be true).
>
> Most of this is just to bill for "slack time", and it extends far beyond
> this, telecom USA did that 1010220 or whatever, 99 cents for the first
> 20 minutes, 5 cents there after. Well if you do the math, its 5 cents
> for each minute, unless you hang up before 20. They relied on the slack
> time for profit (this plan is many years old, it may be cheaper now).
>
>
> >
> --
> Trixter http://www.0xdecafbad.com

Bret McDanel
> Belfast +44 28 9099 6461 US +1 516 687 5200
> http://www.trxtel.com the phone company that pays you!
>
>
> _______________________________________________
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>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>

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>

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Thursday, March 27, 2008

Re: [asterisk-biz] ITSP Billing Increments....?

On Thu, 2008-03-27 at 22:56 -0700, Douglas Garstang wrote:
> So, I have a general question.
>
> What billing increments do ITSP's who terminate SIP->PSTN normally
> bill in?
>
> We've been using carriers that bill 1/1, and the quality hasn't been
> too good, lots of 503's etc. We just started testing with someone else
> that offers much better ASR's (they failover internally between qwest,
> global crossing etc), but they bill 30/6. According to THEM, only
> companies that can't offer top quality routes bill 1/1.
>
> What's the deal here? Is this true?

First, does it really matter? How many calls do you make that are less
than 30 seconds? Will the 6 second increment really affect your total
price? If the new price is higher, is the quality worth it?


I am unsure if there is a relation between billing increments and
quality, but ok.

As for what a provider bills at that is up to the provider itself. This
may be passed on from the carriers they are using to terminate calls.
Generally on the PSTN (at least in the US) you will see a higher first
minute and lower additional minutes. This is because to set up the call
a few queries have to be made which cost the carriers money. Even if
the phone isnt answered there are these costs. So what they do is
charge a higher first minute to recover these costs and when the phone
is never answered they write that off into other costs which is paid for
in part by that first minute and each additional minute.

Now most residential service in the US is billed in whole minute
increments while many business plans are 6 second increments. Really
there isnt any reason for them to not be each second increments.

For short duration calls a higher billing increment is more profitable,
which means that other costs can be absorbed by that.

Since few PSTN based carriers offer 1/1 billing (I think this is
changing though) on most plans it may be true that you cant do 1/1 with
quality carriers as a general rule. I do not however believe it to be
an absolute rule, and again there is cost vs quality.

Its very likely that their carriers are doing 30/6 or 6/6 to them, which
are (or at least were) very common business billing plans. I dont know
why they did 6 seconds, the carriers claimed it had something to do with
the way that duration was calculated over a decade ago, that the systems
would not let them do 1/1. I didnt believe it then, and certainly now
its not true, or if it is there is no real reason for it to be true
(intercarrier settlements are often enough millisecond resolution
showing that it doesnt have to be true).

Most of this is just to bill for "slack time", and it extends far beyond
this, telecom USA did that 1010220 or whatever, 99 cents for the first
20 minutes, 5 cents there after. Well if you do the math, its 5 cents
for each minute, unless you hang up before 20. They relied on the slack
time for profit (this plan is many years old, it may be cheaper now).


>
--
Trixter http://www.0xdecafbad.com

Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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[asterisk-biz] ITSP Billing Increments....?

So, I have a general question.

What billing increments do ITSP's who terminate SIP->PSTN normally bill in?

We've been using carriers that bill 1/1, and the quality hasn't been too good, lots of 503's etc. We just started testing with someone else that offers much better ASR's (they failover internally between qwest, global crossing etc), but they bill 30/6. According to THEM, only companies that can't offer top quality routes bill 1/1.

What's the deal here? Is this true?

Thanks,
Doug.



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[asterisk-biz] Ecuador DIDs

We have a need for 2000 DIDs in Ecuador. Please contact me off the list.

Thanks

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Re: [asterisk-biz] activex softphone

We quit using IAX years ago.

----- Original Message -----
From: "Steve Totaro" <stotaro@totarotechnologies.com>
To: "Commercial and Business-Oriented Asterisk Discussion"
<asterisk-biz@lists.digium.com>
Sent: Thursday, March 27, 2008 4:55 PM
Subject: Re: [asterisk-biz] activex softphone


>
> The army is the people that figure it out on their own or hire me to
> consult their "audio issues". Peek around, check for other
> possibilities and then switch them from IAX to SIP and, boom, no more
> audio issues.
>
> Not sure what qualifies an army but does hundreds count?
>
> Thanks,
> Steve Totaro
>
> On Thu, Mar 27, 2008 at 3:19 PM, Dean Collins <Dean@cognation.net> wrote:
>> Says you and who's army steve :)
>>
>>
>>
>> Regards,
>>
>> Dean Collins
>> Cognation Pty Ltd
>> dean@cognation.net
>> +1-212-203-4357
>> +61-2-9016-5642 (Sydney in-dial).
>>
>>
>>
>> > -----Original Message-----
>> > From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-
>> > bounces@lists.digium.com] On Behalf Of Steve Totaro
>> > Sent: Thursday, 27 March 2008 2:38 PM
>> > To: Commercial and Business-Oriented Asterisk Discussion
>>
>>
>> > Subject: Re: [asterisk-biz] activex softphone
>> >
>> > On Thu, Mar 27, 2008 at 1:13 PM, Tim Panton <thp@westhawk.co.uk>
>> wrote:
>> > >
>> > > On 21 Mar 2008, at 10:38, Andor Czafik (Akakiko) wrote:
>> > >
>> > > > Hi!
>> > > >
>> > > > I need control (answer, and call) sip phone from web browser, and
>> the
>> > > > best is, (i think) the activex softphone.
>> > > > What is the best, and cheaper(or free) activex softphone?
>> > > > Thanks
>> > > > Andor
>> > > >
>> > > > ________________
>> > >
>> > > Does it have to be SIP ? Would IAX do ?
>> > > Do you care about the other 25% of users who use Firefox or Macs?
>> > > Will your users agree to install an activeX control ?
>> > >
>> > > Tim.
>> > >
>> > >

www.phonefromhere.com
>> > >
>> >
>> >
>> > IAX is junk. Never figured out what causes the quality to be so poor,
>> > maybe it is simultaneous usage, maybe it is trunking enabled, maybe it
>> > is just junk.
>> >
>> > It seems to work OK for single calls to boxes taking single calls but
>> > beyond that, the audio is so bad it is not even worth the time trying
>> > to figure it out.
>> >
>> > SIP still rules.
>> >
>> > Thanks,
>> > Steve Totaro
>> >
>>
>>
>> > _______________________________________________
>> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
>> >
>> > 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--
>>
>> 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--
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>

http://lists.digium.com/mailman/listinfo/asterisk-biz

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Re: [asterisk-biz] Pulver media in trouble ?

Not surprising when VON really became a

On Thu, Mar 27, 2008 at 11:39 AM, zoa <zoachien@securax.org> wrote:
> http://www.ipcom-insights.com/blog/marc/default.aspx
>

Not surprising when VON really became an Asterisk advertising agency.

Thanks,
Steve Totaro

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Re: [asterisk-biz] activex softphone

The army is the people that figure it out on their own or hire me to
consult their "audio issues". Peek around, check for other
possibilities and then switch them from IAX to SIP and, boom, no more
audio issues.

Not sure what qualifies an army but does hundreds count?

Thanks,
Steve Totaro

On Thu, Mar 27, 2008 at 3:19 PM, Dean Collins <Dean@cognation.net> wrote:
> Says you and who's army steve :)
>
>
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> dean@cognation.net
> +1-212-203-4357
> +61-2-9016-5642 (Sydney in-dial).
>
>
>
> > -----Original Message-----
> > From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-
> > bounces@lists.digium.com] On Behalf Of Steve Totaro
> > Sent: Thursday, 27 March 2008 2:38 PM
> > To: Commercial and Business-Oriented Asterisk Discussion
>
>
> > Subject: Re: [asterisk-biz] activex softphone
> >
> > On Thu, Mar 27, 2008 at 1:13 PM, Tim Panton <thp@westhawk.co.uk>
> wrote:
> > >
> > > On 21 Mar 2008, at 10:38, Andor Czafik (Akakiko) wrote:
> > >
> > > > Hi!
> > > >
> > > > I need control (answer, and call) sip phone from web browser, and
> the
> > > > best is, (i think) the activex softphone.
> > > > What is the best, and cheaper(or free) activex softphone?
> > > > Thanks
> > > > Andor
> > > >
> > > > ________________
> > >
> > > Does it have to be SIP ? Would IAX do ?
> > > Do you care about the other 25% of users who use Firefox or Macs?
> > > Will your users agree to install an activeX control ?
> > >
> > > Tim.
> > >
> > >

www.phonefromhere.com
> > >
> >
> >
> > IAX is junk. Never figured out what causes the quality to be so poor,
> > maybe it is simultaneous usage, maybe it is trunking enabled, maybe it
> > is just junk.
> >
> > It seems to work OK for single calls to boxes taking single calls but
> > beyond that, the audio is so bad it is not even worth the time trying
> > to figure it out.
> >
> > SIP still rules.
> >
> > Thanks,
> > Steve Totaro
> >
>
>
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > 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--
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>

http://lists.digium.com/mailman/listinfo/asterisk-biz
>

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Re: [asterisk-biz] Looking for solutions.

Jay Call me - 714-228-5410 - Bart

-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com
[mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Jay Kordic
Sent: Thursday, March 27, 2008 9:34 AM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: Re: [asterisk-biz] Looking for solutions.

Nitzan,
I can help you with all of your needs using a few of my carrier partners.

1. Free DID's through some CLEC partners with compensation.
2. Toll free termination(8xx) compensation.
3.A large DID footprint with DID's priced in your price range covering
entire USA (Verizon Footprint).
Please contact me for more information.

Regards,

Jay Kordic
The Horizon Group
Wholesale VOIP/TDM routes/Wholesale IP Bandwidth
1-951-744-9220
1-515-322-0273(fax)
MSN IM- Jaykordic@hotmail.com

March Madness specials
1-ALASKA TERMINATION as low as .0105minute.

2-USA termination offered with and without ANI delivery-ask me about the
many carrier products available.Rates as low as .004/minute for virtual
ANI/CLI delivery,10 million minute minimum per month for this rate.

3-Internet bandwidth from USA Tier 1 providers for Fast E/Gig E as low as
$14/mb.Includes XO,GX,L3,Qwest,UUNet and more.
What services do we provide?

NEED COLOCATION ? Ask me about it.
AVAILABLE CITIES
Atlanta, GA
Charlotte, NC
Chicago, IL - 350 E. Cermak Road
Chicago, IL - 600 S. Federal Street
Dallas, TX - 2323 Bryan Street
Dallas, TX - 8435 Stemmons Fwy
Los Angeles, CA
Miami, FL
NYC - 60 Hudson Street
NYC - 111 Eighth Avenue
Phoenix, AZ
San Francisco, CA
Santa Clara, CA
Weehawken, NJ
-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com
[mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Nitzan Kon
Sent: Wednesday, March 26, 2008 9:58 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: [asterisk-biz] Looking for solutions.

Since it didn't go through the first time.. I'm sending this again:

Hi list!

We are looking for solutions and providers for several different
things:

1. Looking for CLECs that can offer us free DID's. All area codes are
good. If you can offer per-minute kickbacks that is great, but will
take numbers without the kickbacks too.

2. Looking for Toll-Free termination with kickbacks.

3. Need providers with decent local DID origination pricing for all
area codes. Our target range is $1-2 per DID, but we require at least
1000 free incoming minutes.

Most of our customers are residential, with some light commercial.

Looking forward to hearing from you!

--
Nitzan Kon, CEO
Future Nine Corporation
www.future-nine.com

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[WhiteListed]


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Re: [asterisk-biz] activex softphone

Says you and who's army steve :)

Regards,

Dean Collins
Cognation Pty Ltd
dean@cognation.net
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).


> -----Original Message-----
> From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-
> bounces@lists.digium.com] On Behalf Of Steve Totaro
> Sent: Thursday, 27 March 2008 2:38 PM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [asterisk-biz] activex softphone
>
> On Thu, Mar 27, 2008 at 1:13 PM, Tim Panton <thp@westhawk.co.uk>
wrote:
> >
> > On 21 Mar 2008, at 10:38, Andor Czafik (Akakiko) wrote:
> >
> > > Hi!
> > >
> > > I need control (answer, and call) sip phone from web browser, and
the
> > > best is, (i think) the activex softphone.
> > > What is the best, and cheaper(or free) activex softphone?
> > > Thanks
> > > Andor
> > >
> > > ________________
> >
> > Does it have to be SIP ? Would IAX do ?
> > Do you care about the other 25% of users who use Firefox or Macs?
> > Will your users agree to install an activeX control ?
> >
> > Tim.
> >
> >

www.phonefromhere.com
> >
>
>
> IAX is junk. Never figured out what causes the quality to be so poor,
> maybe it is simultaneous usage, maybe it is trunking enabled, maybe it
> is just junk.
>
> It seems to work OK for single calls to boxes taking single calls but
> beyond that, the audio is so bad it is not even worth the time trying
> to figure it out.
>
> SIP still rules.
>
> Thanks,
> Steve Totaro
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> 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--

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To UNSUBSCRIBE or update options visit:

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