[asterisk-users] What are the various models of DID providers

[randulo]

Hi,

Inspired by a recent rant about one particular provider, I am getting
very curious about something I've never mastered. I'd like someone to
explain this here or at least post a link or two that can educate me
and probably countless others who have no knowledge in this area. I'm
sure there are several of you reading this that know all about the
subject.

What are the various business models of these providers, in particular
where are they on the food chain of the DID or trunks they offer?

For example, I have accounts with several well-known providers of SIP,
IAX trunks, hosted pbx and DID. Each of these is located in a
different area, and I would assume they have different peering and
rates they pay to their upstreams. Without naming names, could someone
tackle this? It might help people know what they are getting into when
the open an account.

What are the best *types* of companies for each category: asterisk
testing and home use, small business, larger business, General
Motors...

tia,

/r

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Re: [asterisk-users] a zaptel problem

[fidibus83]

I’m a newbie in Zaptel or Asterisk.

 

What schould I do know?

 

  _  

Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Danny
Nicholas
Gesendet: Montag, 12. Januar 2009 18:10
An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: Re: [asterisk-users] a zaptel problem

 

RED is just off-hook or unavailable (at least in my shop).

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of fidibus83
Sent: Monday, January 12, 2009 11:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] a zaptel problem

 

Hello,

 

I have a problem with zaptel. I hope you can help me.

 

I installed and configure zaptel.

 

ZAPTEL.CONF

 

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31

dchan=16

loadzone = de

defaultzone=de

 

 

But the output of cat /proc/zaptel/*

Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) HDB3/CCS/CRC4
RECOVERING

 

   1 WCT1/0/1 Clear (In use) RED

   2 WCT1/0/2 Clear (In use) RED

   3 WCT1/0/3 Clear (In use) RED

   4 WCT1/0/4 Clear (In use) RED

   5 WCT1/0/5 Clear (In use) RED

   6 WCT1/0/6 Clear (In use) RED

   7 WCT1/0/7 Clear (In use) RED

   8 WCT1/0/8 Clear (In use) RED

   9 WCT1/0/9 Clear (In use) RED

  10 WCT1/0/10 Clear (In use) RED

  11 WCT1/0/11 Clear (In use) RED

  12 WCT1/0/12 Clear (In use) RED

  13 WCT1/0/13 Clear (In use) RED

  14 WCT1/0/14 Clear (In use) RED

  15 WCT1/0/15 Clear (In use) RED

  16 WCT1/0/16 HDLCFCS (In use) RED

  17 WCT1/0/17 Clear (In use) RED

  18 WCT1/0/18 Clear (In use) RED

  19 WCT1/0/19 Clear (In use) RED

  20 WCT1/0/20 Clear (In use) RED

  21 WCT1/0/21 Clear (In use) RED

  22 WCT1/0/22 Clear (In use) RED

  23 WCT1/0/23 Clear (In use) RED

  24 WCT1/0/24 Clear (In use) RED

  25 WCT1/0/25 Clear (In use) RED

  26 WCT1/0/26 Clear (In use) RED

  27 WCT1/0/27 Clear (In use) RD

  28 WCT1/0/28 Clear (In use) RED

  29 WCT1/0/29 Clear (In use) RED

  30 WCT1/0/30 Clear (In use) RED

  31 WCT1/0/31 Clear (In use) RED

 

Why is it RED? zttool shows no alarms (ok) and ztfcg -vv shows that 31
Channels to configure.

 

Thanks!

Best Regards, fidibus 

 
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[asterisk-users] Dahdi caused Kernel to segfault

[Thomas Kenyon]

Yesterday, a low-duty production server that I maintain core-dumped. At 
the time there were only around 2 calls going through it.

The strace on the screen made it look like it was caused by Dahdi.

The machine is running

asterisk-1.6.0.3
dahdi-linux-2.1.0.3
dahdi-tools-2.1.0.2
asterisk-addons-1.6.0

Kernel version 2.6.28

There is a genuine TDM400P (populated witrh 2xFXO cards and 2xFXS cards.

Has anyone had a similar issue?

This has only happened once, but I am a bit worried.


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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

Hi Randulo,

I think this topic is probably more appropriate for asterisk-biz, as was 
the aforementioned rant about one particular DID provider.  But, 
whatever - it is what it is.

I assume that by "DID providers" you are referring to "origination" - 
that is, picking up calls on PSTN numbers and converting them to VoIP 
media and signaling and sending them to someone who wants to get numbers 
that ordinary PSTN users can call on a VoIP system of some kind.  The 
reason for the disambiguation is that many "DID providers" also provide 
"termination" - that is, the delivery of calls from VoIP into the PSTN. 
  There are also many companies that specialise in only origination or 
termination.  The two are closely related from a technical perspective 
but are characterised by rather different economics.

At the end of the day--on a technical and a regulatory level--telephone 
numbers can only belong to a carrier.  A carrier is a network operator 
that is interconnected with other carriers and operates some form of 
switch, and usually interfaces via SS7 (or CSS7, as it is known outside 
North America) to the other carriers that they connect to.

(Aside/digression about carriers:

Of course, there are different types of carriers, depending on the 
jurisdiction.

In the US, there are - broadly speaking - two different types: 
"incumbents" and "competitive" carriers involved in local service. 
Incumbents are either Bell system entities that were divested from the 
former AT&T monopoly in 1984 when AT&T was ordered to break itself up by 
the federal government, or various local-yokel independent telephone 
companies that were never acquired by AT&T during the 20th century (as 
well as various types of conglomerates that have bought some of these 
independents before, or since divestiture).  The latter type of 
incumbent is usually in small towns and/or rural areas, whereas the 
former is prevalent in metropolitan areas.

The defining feature of an incumbent is that it tends to own the 
physical plant related to local telephone service delivery in a given 
area -- copper, fiber, central offices ("telephone exchanges"), remote 
terminals, junction boxes, conduit, and so on.   That's why it's an 
"incumbent."

Examples of incumbents in the US include the former BellSouth (now 
AT&T), Ameritech, Qwest, Southwestern Bell (now AT&T), Verizon, GTE (now 
Verizon), and so on.  Independent incumbents include something like 
Ellijay Telephone Company here in Georgia, or Windstream (formerly 
Alltel).  This space has undergone a dizzying array of consolidation in 
the postmillenial years, so keeping accurate track of who is who even 
for pedagogical purposes is difficult.

The Telecommunications Act of 1996 created "local loop" competition in 
the US and introduced the category of "competitive" carrier, or a CLEC 
(Competitive Local Exchange Carrier).  These are carriers that can 
interconnect with the incumbent (and in fact, the incumbent is legally 
required to interconnect with them) and have the right to lease certain 
parts of the incumbent's infrastructure at regulated rates in order to 
provide subscriber services - this pricing and resale discipline is 
known as UNE (Unbundled Network Element) in the parlance.  For example, 
a CLEC here in Atlanta in former BellSouth territory (now AT&T) connects 
their network to BellSouth and can rent the copper going back to my 
residence from BellSouth and generate all the services, features and 
routing from its own equipment and use BellSouth's plant to reach me 
over the "last mile."  CLECs can do other things as well;  they have 
various rights-of-way that let them build private networks across 
conduits in public spaces, they can lease dark fiber laid by electrical 
and gas utilities, etc.  But the defining feature of a CLEC is that they 
don't own the existing physical plant in place before, although they are 
welcome to overlay their own - in fact, that was very much the point of 
the Telecommunications Act.

Most CLECs are small, but some are quite large and have a regional, 
national and even international footprint.  Examples of the large ones 
include Level3, Global Crossing, XO, McLeod USA, Paetec, Nuvox, etc. -- 
these network operators all have CLEC status in many different 
incumbents' operating areas, if not necessarily all of them.

Some CLECs neither do UNE nor really build networks nor lease anything, 
but exist for some specialised purpose to reap some economic or 
logistical advantage, like supporting the back side of a VoIP product or 
providing dedicated private transport between various large 
interconnection / peering points.  There are many different niches for 
the sort of thing that they are.  Nor does a CLEC have to have an 
imposing physical presence;  it is quite possible, with the right 
equipment, to stuff a fully operational CLEC into half a cabinet in a 
data center.  But at a minimum, a CLEC must run *some* kind of switch 
and interconnect wit

Re: [asterisk-users] Dahdi caused Kernel to segfault

[Benoit]

Personnaly, i had recently encountered  a global machine check exception
with
two cards (TE220p and B410) and many kernel panic with mISDN (mostly if
i tried to unload it).

Dahdi still hasn't failed me (directly)

Thomas Kenyon a écrit :
> Yesterday, a low-duty production server that I maintain core-dumped. At 
> the time there were only around 2 calls going through it.
>
> The strace on the screen made it look like it was caused by Dahdi.
>
> The machine is running
>
> asterisk-1.6.0.3
> dahdi-linux-2.1.0.3
> dahdi-tools-2.1.0.2
> asterisk-addons-1.6.0
>
> Kernel version 2.6.28
>
> There is a genuine TDM400P (populated witrh 2xFXO cards and 2xFXS cards.
>
> Has anyone had a similar issue?
>
> This has only happened once, but I am a bit worried.
>
>
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   



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Re: [asterisk-users] recommendation for German sound files

[Klaus Darilion]

Philipp Kempgen schrieb:
> === Amooma ===
> 
> * http://www.amooma.de/asterisk/sprachbausteine/#prompts-tts
> These files are generated by our web-based text-to-speech engine.
> Pros: If you need additional custom prompts, just go to
> http://www.amooma.de/tts/ and generate them and the voice will
> match.

Hi Philipp!


What is the license of these sound files generated on the website?

thanks
klaus

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Re: [asterisk-users] a zaptel problem

[Steve Howes]

http://www.voip-info.org

Read all that.

On 13 Jan 2009, at 08:51, fidibus83 wrote:

> I’m a newbie in Zaptel or Asterisk.
>
> What schould I do know?
>
> Von: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com 
> ] Im Auftrag von Danny Nicholas
> Gesendet: Montag, 12. Januar 2009 18:10
> An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Betreff: Re: [asterisk-users] a zaptel problem
>
> RED is just off-hook or unavailable (at least in my shop).
>
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com 
> ] On Behalf Of fidibus83
> Sent: Monday, January 12, 2009 11:04 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] a zaptel problem
>
> Hello,
>
> I have a problem with zaptel. I hope you can help me.
>
> I installed and configure zaptel.
>
> ZAPTEL.CONF
>
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
> loadzone = de
> defaultzone=de
>
>
> But the output of cat /proc/zaptel/*
> Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) HDB3/ 
> CCS/CRC4 RECOVERING
>
>1 WCT1/0/1 Clear (In use) RED
>2 WCT1/0/2 Clear (In use) RED
>3 WCT1/0/3 Clear (In use) RED
>4 WCT1/0/4 Clear (In use) RED
>5 WCT1/0/5 Clear (In use) RED
>6 WCT1/0/6 Clear (In use) RED
>7 WCT1/0/7 Clear (In use) RED
>8 WCT1/0/8 Clear (In use) RED
>9 WCT1/0/9 Clear (In use) RED
>   10 WCT1/0/10 Clear (In use) RED
>   11 WCT1/0/11 Clear (In use) RED
>   12 WCT1/0/12 Clear (In use) RED
>   13 WCT1/0/13 Clear (In use) RED
>   14 WCT1/0/14 Clear (In use) RED
>   15 WCT1/0/15 Clear (In use) RED
>   16 WCT1/0/16 HDLCFCS (In use) RED
>   17 WCT1/0/17 Clear (In use) RED
>   18 WCT1/0/18 Clear (In use) RED
>   19 WCT1/0/19 Clear (In use) RED
>   20 WCT1/0/20 Clear (In use) RED
>   21 WCT1/0/21 Clear (In use) RED
>   22 WCT1/0/22 Clear (In use) RED
>   23 WCT1/0/23 Clear (In use) RED
>   24 WCT1/0/24 Clear (In use) RED
>   25 WCT1/0/25 Clear (In use) RED
>   26 WCT1/0/26 Clear (In use) RED
>   27 WCT1/0/27 Clear (In use) RD
>   28 WCT1/0/28 Clear (In use) RED
>   29 WCT1/0/29 Clear (In use) RED
>   30 WCT1/0/30 Clear (In use) RED
>   31 WCT1/0/31 Clear (In use) RED
>
> Why is it RED? zttool shows no alarms (ok) and ztfcg -vv shows that  
> 31 Channels to configure.
>
> Thanks!
> Best Regards, fidibus
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] What are the various models of DID providers

[randulo]

On Tue, Jan 13, 2009 at 10:49 AM, Alex Balashov
 wrote:
> I think this topic is probably more appropriate for asterisk-biz, as was
> the aforementioned rant about one particular DID provider.  But,
> whatever - it is what it is.

Alex, thanks for the excellent explanations! Exactly what I was hoping
for. Yes, in fact I should have said "origination and termination".

I posted here on purpose because I think both users and budding devels
need to better understand this. I did not mean for it to develop into
a pissing contest or further rants, which is why I thought it might be
best not to mention names except for examples in the food chain.

/r

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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

I will add that the model of the VoIP ITSP (be it a DID provider or a 
termination provider) is inherently a squeezed one.

The original niche of these providers was to simply provide VoIP in the 
first place;  established carriers were used to dealing with TDM (hard, 
synchronous circuits using ISDN or SS7) and had a lot invested in 
equipment and plant to do traditional telco stuff.  So, someone had to 
be doing the TDM<->VoIP conversion work by buying the equipment and 
getting the facilities in place.  There was a point where if you didn't 
want to order physical PRIs yourself and buy equipment to terminate 
them, but instead just wanted straight VoIP right to you, you 
practically had to go to a VoIP provider to get that.  It took the 
carriers a while to upgrade their networks to the sorts of softswitch 
platforms that can generate SIP trunks out to you.  Many carriers still 
can't do it.  Big fish have very big, lumbering, slow-moving processes.

But the purpose of ITSPs has now shifted more to providing cost and 
scalability advantages rather than exclusively infrastructural and 
technological ones.  Maybe you can get SIP trunks from carriers now, but 
for them to deal with you or offer you any sort of attractive pricing, 
you have to be pushing a lot of minutes and buying a lot of numbers.

This will only get more true as carriers develop business processes and 
technology platforms that make it more economical for them to 
accommodate a wider range of customer size directly.  It's sort of the 
same thing that happened to independent ISPs providing dialup Internet 
access when the telcos themselves got into that game.

But there are places where ITSPs can offer a lot of value, even for 
large amounts of traffic.  Sometimes there's a pricing advantage, other 
times it's good service and solid backoffice tools and processes that 
streamline the experience for the customer.  It still takes some work to 
enter into a wholesale agreement with a carrier, even though it's gotten 
a lot easier now as evidenced by the mushrooming number of small 
providers.  Big carriers are very slow-moving organisations, so it often 
helps to have an ITSP on your side dealing with provisioning processes 
and change orders in a way that insulates you from the responsibility. 
Good ITSPs maintain standing DID inventories and make it easy for you to 
just point and click yourself another number, configure many different 
destination gateways for those numbers, etc, etc.  These are all things 
underlying carriers are in a worse position to provide, especially with 
anything that might be described as speed or ease.  It can be very 
beneficial to have someone aggregate that stuff and package it up into a 
product and service delivery platform oriented toward ordinary users.

So, like I said, it's not too different from the reasons for which 
distribution channels exist in other types of supply chains in other 
industries.


Alex Balashov wrote:

> Hi Randulo,
> 
> I think this topic is probably more appropriate for asterisk-biz, as was 
> the aforementioned rant about one particular DID provider.  But, 
> whatever - it is what it is.
> 
> I assume that by "DID providers" you are referring to "origination" - 
> that is, picking up calls on PSTN numbers and converting them to VoIP 
> media and signaling and sending them to someone who wants to get numbers 
> that ordinary PSTN users can call on a VoIP system of some kind.  The 
> reason for the disambiguation is that many "DID providers" also provide 
> "termination" - that is, the delivery of calls from VoIP into the PSTN. 
>   There are also many companies that specialise in only origination or 
> termination.  The two are closely related from a technical perspective 
> but are characterised by rather different economics.
> 
> At the end of the day--on a technical and a regulatory level--telephone 
> numbers can only belong to a carrier.  A carrier is a network operator 
> that is interconnected with other carriers and operates some form of 
> switch, and usually interfaces via SS7 (or CSS7, as it is known outside 
> North America) to the other carriers that they connect to.
> 
> (Aside/digression about carriers:
> 
> Of course, there are different types of carriers, depending on the 
> jurisdiction.
> 
> In the US, there are - broadly speaking - two different types: 
> "incumbents" and "competitive" carriers involved in local service. 
> Incumbents are either Bell system entities that were divested from the 
> former AT&T monopoly in 1984 when AT&T was ordered to break itself up by 
> the federal government, or various local-yokel independent telephone 
> companies that were never acquired by AT&T during the 20th century (as 
> well as various types of conglomerates that have bought some of these 
> independents before, or since divestiture).  The latter type of 
> incumbent is usually in small towns and/or rural areas, whereas the 
> former is prevalent in metropolitan area

Re: [asterisk-users] What are the various models of DID providers

[randulo]

On Tue, Jan 13, 2009 at 11:22 AM, Alex Balashov
 wrote:
> The original niche of these providers was to simply provide VoIP in the
> first place;  established carriers were used to dealing with TDM (hard,
> synchronous circuits using ISDN or SS7) and had a lot invested in
> equipment and plant to do traditional telco stuff.  So, someone had to
> be doing the TDM<->VoIP conversion work by buying the equipment and

When you think about the learning curve of the average asterisk
beginner, the picture painted for them is "Become your own telco!" and
we all know that's not exactly accurate. For the small asterisk
install it's much more accurate to say "Get an enterprise-class pbx
free (if you don't mind months of studying)."

The biggest challenge for anyone who will NOT be dealing with the
carriers or large wholesalers is the inherent instability of the
market for affordable ITSP. Over the past few years, many of these
have fallen by the wayside or gotten so bad users began to jump ship
on their own. Expect one or more announcements VERY SOON from names
you've heard about.

My best experience is from our DSL and ITSP. I can be talking to a
human being immediately and a service tech who knows what VoIP is
within a minute. They're fantastic, but I see they are about to go
public and be on the NASDAQ. I hope that doesn't compromise their
great service.

> This will only get more true as carriers develop business processes and
> technology platforms that make it more economical for them to
> accommodate a wider range of customer size directly.  It's sort of the
> same thing that happened to independent ISPs providing dialup Internet
> access when the telcos themselves got into that game.

You think someday AT&T will have a web site where you can sign up for
termination/origination? I guess the airlines are able to sell flights
that way. Customer service would be a nightmare though. Welcome to
"Your call is important to us" land.

/r

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Re: [asterisk-users] a zaptel problem

[Tzafrir Cohen]

On Tue, Jan 13, 2009 at 10:18:30AM +, Steve Howes wrote:
> http://www.voip-info.org
> 
> Read all that.

Great "RTFM" answer. Let me be more specific: what does "RECOVERING"
mean?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

randulo wrote:

> Alex, thanks for the excellent explanations! Exactly what I was hoping
> for. 

Good!  Glad to help.

> Yes, in fact I should have said "origination and termination".

Ah.  Termination is a somewhat different game than origination.  From a 
technical point of view, most of what I said about origination also 
applies, and there's the same kind of arbitrage of underlying carrier 
pricing going on in the exact same way.

But all other things being equal, origination is an easier and better 
business model for an ITSP to be in.  That's why you'll notice quite a 
bit of origination providers who don't want to touch termination with a 
ten foot pole.

It has to do with costs, exposure and regulation.  In the US, at least, 
carriers *like* inbound traffic to be sent to them by other carriers. 
If you get a SIP trunk from Level3 and get some numbers from them, they 
take considerable delight in customers of other carriers calling you. 
Aside from using resources, it doesn't really cost them anything.  In 
fact, carriers actually pay each other to terminate calls amongst 
themselves;  this is known as "reciprocal compensation" or "intercarrier 
settlement."   When carrier A sends a call to carrier B, carrier B gets 
to bill carrier A a fairly infinitesimal but still nonzero rate.  The 
idea behind that is that the resources consumed do have a nontrivial 
cost associated with them, and unlike carrier A, carrier B has to accept 
the call and has no control over the volume.

Reciprocal compensation is a very complicated subject, and indeed, 
probably the single most broken aspect of telecom law in the US right 
now.  There are lots of things that are exempt from it with varying 
degrees of practicality and enforceability, such as VoIP traffic 
(defined as traffic that originates from VoIP customer premise 
equipment) and ISP-bound modem traffic[1].  Not everyone has to bill the 
same rate;  metropolitan incumbents mostly do, but rural carriers 
fixed-line carriers get to charge special - often much higher - rates in 
consideration of their low-ROI "rural" build-out characteristics[2]. 
It's an effective subsidy.

The point is that if you are doing origination as an ITSP, it's pretty 
standard to get unlimited inbound usage per channel (call) for a flat 
rate, which is something you can then in turn offer to your customers. 
It cuts out the exposure and makes for an attractive product, since 
people don't like paying usage (you don't pay it with a conventional 
landline).

If you're doing termination, though, you have to face this problem from 
the other side.  The carrier *will* bill you for usage, so at the very 
least, you have to make sure you aren't out money due to failures of 
billing reconciliation, rounding, etc. which keeps you awake at night 
worrying about the accuracy of your CDRs (Call Detail Records) relative 
to what the carrier sees.  Long-distance competition is a lot older than 
local loop competition (it was part of the Modification of Final 
Judgment that broke up AT&T in 1984;  AT&T's divested Regional Bell 
Operating Companies (RBOCs) would be allowed to retain their local loop 
monopoly in exchange for allowing competition in long-distance hauling), 
which helped domestic LD prices collapse over time along with the advent 
of CLECs and VoIP.  As a result, the margins are very, very thin; 
slight billing and rating errors mean you can be out a lot of money if 
you collect just a little less from your users than your carrier bills 
you.  Plus there's fraud to worry about.

It's actually a lot worse than that for the reasons I mentioned above; 
depending on which carrier the call is ultimately going to, *your* 
proximate termination carrier faces different access costs.  As a 
result, the country is split up into various "tiers" of traffic. 
Terminating traffic into RBOCs costs one thing, into mobile providers 
another, rural independents a third, and so on and so forth.  That means 
you have to be aware of the derivatives of these cost differences and 
bill your customers accordingly, and if you want to hear stories about 
how hard that can be, talk to some termination guys.

It is possible to get a "blended" rate based on an average derived from 
certain assumptions stipulated in the contract about the composition of 
your traffic destinations, but that has a number of problems.  You can 
be thrown off-guard by violations of the blend agreement if your carrier 
decides to pursue you for unexpected higher costs, and it doesn't get 
you the most competitive rate to offer to your customers for traffic 
destinations that *are* cheap to call.  Plus, with higher volumes, you 
get screwed because you can bill your customers one rate and make more 
money off the arbitrage to cheaper destinations while eating a loss on 
your gross margins on the more expensive ones.  So, higher traffic 
volumes tend to send an ITSP toward a "decked" or "tiered" arrangement 
with their carrier (by rate ce

Re: [asterisk-users] a zaptel problem

[David fire]

RED means the cable is unpluged or misconfigured.
you also need to configure zapata.conf in /etc/asterisk
David

2009/1/12 fidibus83 

>  Hello,
>
>
>
> I have a problem with zaptel. I hope you can help me.
>
>
>
> I installed and configure zaptel.
>
>
>
> ZAPTEL.CONF
>
>
>
> span=1,1,0,ccs,hdb3,crc4
>
> bchan=1-15,17-31
>
> dchan=16
>
> loadzone = de
>
> defaultzone=de
>
>
>
>
>
> But the output of cat /proc/zaptel/*
>
> Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) HDB3/CCS/CRC4
> RECOVERING
>
>
>
>1 WCT1/0/1 Clear (In use) RED
>
>2 WCT1/0/2 Clear (In use) RED
>
>3 WCT1/0/3 Clear (In use) RED
>
>4 WCT1/0/4 Clear (In use) RED
>
>5 WCT1/0/5 Clear (In use) RED
>
>6 WCT1/0/6 Clear (In use) RED
>
>7 WCT1/0/7 Clear (In use) RED
>
>8 WCT1/0/8 Clear (In use) RED
>
>9 WCT1/0/9 Clear (In use) RED
>
>   10 WCT1/0/10 Clear (In use) RED
>
>   11 WCT1/0/11 Clear (In use) RED
>
>   12 WCT1/0/12 Clear (In use) RED
>
>   13 WCT1/0/13 Clear (In use) RED
>
>   14 WCT1/0/14 Clear (In use) RED
>
>   15 WCT1/0/15 Clear (In use) RED
>
>   16 WCT1/0/16 HDLCFCS (In use) RED
>
>   17 WCT1/0/17 Clear (In use) RED
>
>   18 WCT1/0/18 Clear (In use) RED
>
>   19 WCT1/0/19 Clear (In use) RED
>
>   20 WCT1/0/20 Clear (In use) RED
>
>   21 WCT1/0/21 Clear (In use) RED
>
>   22 WCT1/0/22 Clear (In use) RED
>
>   23 WCT1/0/23 Clear (In use) RED
>
>   24 WCT1/0/24 Clear (In use) RED
>
>   25 WCT1/0/25 Clear (In use) RED
>
>   26 WCT1/0/26 Clear (In use) RED
>
>   27 WCT1/0/27 Clear (In use) RD
>
>   28 WCT1/0/28 Clear (In use) RED
>
>   29 WCT1/0/29 Clear (In use) RED
>
>   30 WCT1/0/30 Clear (In use) RED
>
>   31 WCT1/0/31 Clear (In use) RED
>
>
>
> Why is it RED? zttool shows no alarms (ok) and ztfcg -vv shows that 31
> Channels to configure.
>
>
>
> Thanks!
>
> Best Regards, fidibus
>
>
>
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>
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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

randulo wrote:

> When you think about the learning curve of the average asterisk
> beginner, the picture painted for them is "Become your own telco!" and
> we all know that's not exactly accurate. For the small asterisk
> install it's much more accurate to say "Get an enterprise-class pbx
> free (if you don't mind months of studying)."

I can definitely concur with that.  But misappropriation of Asterisk and 
naive business models arising from it is a whole different topic.  :-)

> The biggest challenge for anyone who will NOT be dealing with the
> carriers or large wholesalers is the inherent instability of the
> market for affordable ITSP. Over the past few years, many of these
> have fallen by the wayside or gotten so bad users began to jump ship
> on their own. Expect one or more announcements VERY SOON from names
> you've heard about.

It's a hard business to be in.  The margins are thin, and carriers will 
go after your biggest and most profitable customers -- assuming those 
customers aren't ingenious enough to try to cut you out on their own.

This type of resale is commonly spoken to by the somewhat pejorative 
term "arbitrage play."  It implies resale without adding a lot (if any) 
real value by people who don't understand PSTN economics or good 
engineering.  Arbitrage-based business models are also characterised 
generally by volatility and unsustainability;  in the long run, the 
market rationalises them away as the offering achieves higher 
penetration and becomes more commoditised.  People start wanting to go 
straight to the farmer to get their wheat.

However, as I said in the previous post, there are legitimate 
opportunities for ITSPs to add lasting long-term value.  To make it 
happen, though, there's got to be more than resale of minutes going on. 
   Innovation of business process and streamlining of cost will prove 
much more important in the long run as a value proposition.  Competing 
on price is just a race to the bottom.  This is especially true given 
the source of a lot of the cost basis in current regulatory conditions, 
which are, to put it mildly, rather CLEC-unfriendly.  The Bush FCC did a 
lot to roll back the pro-small busines gains of TA96--or so goes one 
point of view, anyway.

The other thing to observe is that right now, the voice traffic is still 
fundamentally exchanged through the PSTN -- even among calls between 
VoIP providers.  If I'm on a local ITSP and you call me from Vonage, the 
call is still traversing various IXCs and Bell tandems.  While nobody 
seriously expects the PSTN to go away substantively any time soon, there 
is considerable work being done by the market to find a viable private 
VoIP peering model so that providers can send each other traffic without 
paying the Bells for it at all.  This would represent a rather radical - 
if subtle - departure from the telco business model (monetisation of 
"minutes," calls as sequences of billable events and resource 
acquisition on fixed-bandwidth channel reservations on synchronous 
multiplexed signals) to the Internet business model (flat-rate or 
settlement-free peering and exchange of packets).

Many people thought universal public ENUM would be the ticket, but for 
some reason that hasn't really taken off.  But eventually, ITSPs will 
get smarter and start passing traffic between themselves over pure IP a 
lot more, creating a more serious alternative network overlay and 
delivery model.  As that happens, I think ITSPs will have a much more 
important role to play because they will slowly become the "new" telcos, 
as opposed to just playing resale games.

> My best experience is from our DSL and ITSP. I can be talking to a
> human being immediately and a service tech who knows what VoIP is
> within a minute. They're fantastic, but I see they are about to go
> public and be on the NASDAQ. I hope that doesn't compromise their
> great service.

I know what you mean;  my background is in independent ISP land, back 
when that business model still existed[1].  Independent ISPs generally 
tried to differentiate themselves in marketing with a higher level of 
no-nonsense customer service.

It's tough not to compromise that kind of service;  it's expensive to 
offer and very costly to scale.  Competition in business drives things 
down to a more Pareto-optimal[2] common denominator, wherein to maintain 
your competitive position you get dragged into doing things that are 80% 
good for 80% of customers 80% of the time just like your competitors.

> You think someday AT&T will have a web site where you can sign up for
> termination/origination? I guess the airlines are able to sell flights
> that way. Customer service would be a nightmare though. Welcome to
> "Your call is important to us" land.

I doubt AT&T will, seeing as accelerating or enabling VoIP is not really 
in the incumbents' interest.  But I think the big competitive carriers 
will certainly evolve in that direction.

-- Alex

[1]  I posted some ra

Re: [asterisk-users] What are the various models of DID providers

[Gordon Henderson]

On Tue, 13 Jan 2009, randulo wrote:

> Hi,
>
> Inspired by a recent rant about one particular provider, I am getting
> very curious about something I've never mastered. I'd like someone to
> explain this here or at least post a link or two that can educate me
> and probably countless others who have no knowledge in this area. I'm
> sure there are several of you reading this that know all about the
> subject.
>
> What are the various business models of these providers, in particular
> where are they on the food chain of the DID or trunks they offer?
>
> For example, I have accounts with several well-known providers of SIP,
> IAX trunks, hosted pbx and DID. Each of these is located in a
> different area, and I would assume they have different peering and
> rates they pay to their upstreams. Without naming names, could someone
> tackle this? It might help people know what they are getting into when
> the open an account.
>
> What are the best *types* of companies for each category: asterisk
> testing and home use, small business, larger business, General
> Motors...

"What country are you in?" is always a good start. Remember VoIP is 
global, as are the members of this list.

For example, I'm in the UK and here we have a good selection of both 
inbound and outbound providers, although sometimes it's not obvious where 
they are in the food-chain. (Or who's food-chain they're part of!)

For me, I'm what I might view as a "middle man" in the food-chain, I want 
inbound and outbound providers who's own equipment connects directly to 
the PSTN, or as directly as possible. I don't want someone who I connect 
to via VoIP who then connects via VoIP to someone else who connects to the 
PSTN... (Although that's what I offer my customers... Hmm! However I like 
to think I offer more flexability, support and options than the top-level 
guys... Seems to work for me!)

So in the UK we have a small number of people in the same boat as me - 
connecting to inbound and outbound providers and selling-on these services 
either in the virtual form, or providing hardware and "trunks" (SIP and/or 
IAX) as required. My own value-add is to go one step further and provide 
the PBX (and Internet connectivity, if required) as well. Sort of one-stop 
shop.

There is also a plethora of providers aimed strictly at the residential 
market, and those who are trying to sell (expensive IMO) hosted solutions 
to the SME.

For me, inbound is more the issue than outbound, as I register numbers for 
inbound via the wholesalers who are then the only people who can route 
calls to that number to me, but I can send calls out via more than one 
operator (as long as they allow me to present outgoing CID from other 
operators) If an inbound operator fails, then all those numnbers are lost 
for the time being, so it pays to do a bit of research!

And that's what it boils down to - research. This could be by contacting 
the ITSPs directly, by word of mouth on various forums, etc. and so on. In 
the UK we have the ITSPA - http://www.itspa.org.uk/ and most of the main 
players are members, as well as some of the lesser ones, so it's a good 
place to start.

Gordon

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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

The interesting thing about the UK is that you folks did local loop 
deregulation right, unlike in the US, where it's vicious and toxic and 
is mostly a story about the bewildering multitude of ways in which the 
incumbents screw competitive CLECs.

No, the results of the BT bifurcation isn't perfect, but having even an 
"officially" neutral LoopCo like OpenReach would go a long way in the US.

Gordon Henderson wrote:

> On Tue, 13 Jan 2009, randulo wrote:
> 
>> Hi,
>>
>> Inspired by a recent rant about one particular provider, I am getting
>> very curious about something I've never mastered. I'd like someone to
>> explain this here or at least post a link or two that can educate me
>> and probably countless others who have no knowledge in this area. I'm
>> sure there are several of you reading this that know all about the
>> subject.
>>
>> What are the various business models of these providers, in particular
>> where are they on the food chain of the DID or trunks they offer?
>>
>> For example, I have accounts with several well-known providers of SIP,
>> IAX trunks, hosted pbx and DID. Each of these is located in a
>> different area, and I would assume they have different peering and
>> rates they pay to their upstreams. Without naming names, could someone
>> tackle this? It might help people know what they are getting into when
>> the open an account.
>>
>> What are the best *types* of companies for each category: asterisk
>> testing and home use, small business, larger business, General
>> Motors...
> 
> "What country are you in?" is always a good start. Remember VoIP is 
> global, as are the members of this list.
> 
> For example, I'm in the UK and here we have a good selection of both 
> inbound and outbound providers, although sometimes it's not obvious where 
> they are in the food-chain. (Or who's food-chain they're part of!)
> 
> For me, I'm what I might view as a "middle man" in the food-chain, I want 
> inbound and outbound providers who's own equipment connects directly to 
> the PSTN, or as directly as possible. I don't want someone who I connect 
> to via VoIP who then connects via VoIP to someone else who connects to the 
> PSTN... (Although that's what I offer my customers... Hmm! However I like 
> to think I offer more flexability, support and options than the top-level 
> guys... Seems to work for me!)
> 
> So in the UK we have a small number of people in the same boat as me - 
> connecting to inbound and outbound providers and selling-on these services 
> either in the virtual form, or providing hardware and "trunks" (SIP and/or 
> IAX) as required. My own value-add is to go one step further and provide 
> the PBX (and Internet connectivity, if required) as well. Sort of one-stop 
> shop.
> 
> There is also a plethora of providers aimed strictly at the residential 
> market, and those who are trying to sell (expensive IMO) hosted solutions 
> to the SME.
> 
> For me, inbound is more the issue than outbound, as I register numbers for 
> inbound via the wholesalers who are then the only people who can route 
> calls to that number to me, but I can send calls out via more than one 
> operator (as long as they allow me to present outgoing CID from other 
> operators) If an inbound operator fails, then all those numnbers are lost 
> for the time being, so it pays to do a bit of research!
> 
> And that's what it boils down to - research. This could be by contacting 
> the ITSPs directly, by word of mouth on various forums, etc. and so on. In 
> the UK we have the ITSPA - http://www.itspa.org.uk/ and most of the main 
> players are members, as well as some of the lesser ones, so it's a good 
> place to start.
> 
> Gordon
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] What are the various models of DID providers

[Gordon Henderson]

On Tue, 13 Jan 2009, Alex Balashov wrote:

> The interesting thing about the UK is that you folks did local loop
> deregulation right, unlike in the US, where it's vicious and toxic and
> is mostly a story about the bewildering multitude of ways in which the
> incumbents screw competitive CLECs.

For various values of "right" :)

Number porting is still a PITA.

> No, the results of the BT bifurcation isn't perfect, but having even an
> "officially" neutral LoopCo like OpenReach would go a long way in the US.

Yes. "officially" ... Some would think that in practice, it's not though.

However that's politics for you :)

Gordon


>
> Gordon Henderson wrote:
>
>> On Tue, 13 Jan 2009, randulo wrote:
>>
>>> Hi,
>>>
>>> Inspired by a recent rant about one particular provider, I am getting
>>> very curious about something I've never mastered. I'd like someone to
>>> explain this here or at least post a link or two that can educate me
>>> and probably countless others who have no knowledge in this area. I'm
>>> sure there are several of you reading this that know all about the
>>> subject.
>>>
>>> What are the various business models of these providers, in particular
>>> where are they on the food chain of the DID or trunks they offer?
>>>
>>> For example, I have accounts with several well-known providers of SIP,
>>> IAX trunks, hosted pbx and DID. Each of these is located in a
>>> different area, and I would assume they have different peering and
>>> rates they pay to their upstreams. Without naming names, could someone
>>> tackle this? It might help people know what they are getting into when
>>> the open an account.
>>>
>>> What are the best *types* of companies for each category: asterisk
>>> testing and home use, small business, larger business, General
>>> Motors...
>>
>> "What country are you in?" is always a good start. Remember VoIP is
>> global, as are the members of this list.
>>
>> For example, I'm in the UK and here we have a good selection of both
>> inbound and outbound providers, although sometimes it's not obvious where
>> they are in the food-chain. (Or who's food-chain they're part of!)
>>
>> For me, I'm what I might view as a "middle man" in the food-chain, I want
>> inbound and outbound providers who's own equipment connects directly to
>> the PSTN, or as directly as possible. I don't want someone who I connect
>> to via VoIP who then connects via VoIP to someone else who connects to the
>> PSTN... (Although that's what I offer my customers... Hmm! However I like
>> to think I offer more flexability, support and options than the top-level
>> guys... Seems to work for me!)
>>
>> So in the UK we have a small number of people in the same boat as me -
>> connecting to inbound and outbound providers and selling-on these services
>> either in the virtual form, or providing hardware and "trunks" (SIP and/or
>> IAX) as required. My own value-add is to go one step further and provide
>> the PBX (and Internet connectivity, if required) as well. Sort of one-stop
>> shop.
>>
>> There is also a plethora of providers aimed strictly at the residential
>> market, and those who are trying to sell (expensive IMO) hosted solutions
>> to the SME.
>>
>> For me, inbound is more the issue than outbound, as I register numbers for
>> inbound via the wholesalers who are then the only people who can route
>> calls to that number to me, but I can send calls out via more than one
>> operator (as long as they allow me to present outgoing CID from other
>> operators) If an inbound operator fails, then all those numnbers are lost
>> for the time being, so it pays to do a bit of research!
>>
>> And that's what it boils down to - research. This could be by contacting
>> the ITSPs directly, by word of mouth on various forums, etc. and so on. In
>> the UK we have the ITSPA - http://www.itspa.org.uk/ and most of the main
>> players are members, as well as some of the lesser ones, so it's a good
>> place to start.
>>
>> Gordon
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> -- 
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 237-1775
>
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Re: [asterisk-users] What are the various models of DID providers

[Fred Posner]

I'm really liking Flowroute right now and have switched most of my  
did's to them. As with many of the providers lately, they are pre- 
pay... but the site is nice and payments move quickly. What I really  
like about Flowroute is the no-minimum, diversity of plans, cost, and  
unlimited channel support. For example, on a line I have that's  
residential, I use their 6.95 unlimited incoming 2 channel plan... on  
some business lines where I might need channel flexibility, I use  
unlimited channel 1.2cents/min incoming. Outbound is around less than  
1c/m for USA and pretty decent prices throughout.

I had been using Voicepulse, but suffered more downtime last year than  
I hoped and found that support was a little hard to get a hold of when  
issues occur.

Fred Posner
f...@teamforrest.com

Main:   +1 (212) 937-7844
Direct: +1 (503) 914-0999

www.teamforrest.com

On Jan 13, 2009, at 3:29 AM, randulo wrote:

> Hi,
>
> Inspired by a recent rant about one particular provider, I am getting
> very curious about something I've never mastered. I'd like someone to
> explain this here or at least post a link or two that can educate me
> and probably countless others who have no knowledge in this area. I'm
> sure there are several of you reading this that know all about the
> subject.
>
> What are the various business models of these providers, in particular
> where are they on the food chain of the DID or trunks they offer?
>
> For example, I have accounts with several well-known providers of SIP,
> IAX trunks, hosted pbx and DID. Each of these is located in a
> different area, and I would assume they have different peering and
> rates they pay to their upstreams. Without naming names, could someone
> tackle this? It might help people know what they are getting into when
> the open an account.
>
> What are the best *types* of companies for each category: asterisk
> testing and home use, small business, larger business, General
> Motors...
>
> tia,
>
> /r
>



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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

Gordon Henderson wrote:
> On Tue, 13 Jan 2009, Alex Balashov wrote:
> 
>> The interesting thing about the UK is that you folks did local loop
>> deregulation right, unlike in the US, where it's vicious and toxic and
>> is mostly a story about the bewildering multitude of ways in which the
>> incumbents screw competitive CLECs.
> 
> For various values of "right" :)
> 
> Number porting is still a PITA.

Porting is a pain anywhere.

> 
>> No, the results of the BT bifurcation isn't perfect, but having even an
>> "officially" neutral LoopCo like OpenReach would go a long way in the US.
> 
> Yes. "officially" ... Some would think that in practice, it's not though.

Yeah, but Oftel is pretty serious about at least keeping up the 
appearance of neutrality. Whereas here, nobody even bothers going 
through the motions;  it is rendered explicitly clear that the Federal 
Communications Commission (FCC) is bought and paid for by the incumbent 
RBOCs and that's pretty much all there is to it.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

Managing financial exposure on postpaid can be hard stuff.  Some give up 
on it.

Also, despite the fact that it's a well-known and tiresome boondoggle - 
as well as a fairly moribund loophole - at this point, there seems to be 
no shortage of VoIP outfits blundering their way into the carrier access 
billing arbitrage I described in my post on termination.  The DIDs are 
free and completely unlimited (and indeed, in some cases, they'll even 
pay you a little tiny bit for the usage), but the catch is that they're 
in some really bizarre, rural rate center (area code and exchange, or 
NPA-NXX).

This is them setting up shop with a rural independent and getting a cut 
of the access charges.

Beware, this is in the process of going away, and is generally a really 
stupid idea despite the phenomenal amounts of money that have been made 
with it in times past.

Fred Posner wrote:

> I'm really liking Flowroute right now and have switched most of my  
> did's to them. As with many of the providers lately, they are pre- 
> pay... but the site is nice and payments move quickly. What I really  
> like about Flowroute is the no-minimum, diversity of plans, cost, and  
> unlimited channel support. For example, on a line I have that's  
> residential, I use their 6.95 unlimited incoming 2 channel plan... on  
> some business lines where I might need channel flexibility, I use  
> unlimited channel 1.2cents/min incoming. Outbound is around less than  
> 1c/m for USA and pretty decent prices throughout.
> 
> I had been using Voicepulse, but suffered more downtime last year than  
> I hoped and found that support was a little hard to get a hold of when  
> issues occur.
> 
> Fred Posner
> f...@teamforrest.com
> 
> Main: +1 (212) 937-7844
> Direct:   +1 (503) 914-0999
> 
> www.teamforrest.com
> 
> On Jan 13, 2009, at 3:29 AM, randulo wrote:
> 
>> Hi,
>>
>> Inspired by a recent rant about one particular provider, I am getting
>> very curious about something I've never mastered. I'd like someone to
>> explain this here or at least post a link or two that can educate me
>> and probably countless others who have no knowledge in this area. I'm
>> sure there are several of you reading this that know all about the
>> subject.
>>
>> What are the various business models of these providers, in particular
>> where are they on the food chain of the DID or trunks they offer?
>>
>> For example, I have accounts with several well-known providers of SIP,
>> IAX trunks, hosted pbx and DID. Each of these is located in a
>> different area, and I would assume they have different peering and
>> rates they pay to their upstreams. Without naming names, could someone
>> tackle this? It might help people know what they are getting into when
>> the open an account.
>>
>> What are the best *types* of companies for each category: asterisk
>> testing and home use, small business, larger business, General
>> Motors...
>>
>> tia,
>>
>> /r
>>
> 
> 
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] recommendation for German sound files

[Philipp Kempgen]

Klaus Darilion schrieb:
> Philipp Kempgen schrieb:
>> === Amooma ===
>> 
>> * http://www.amooma.de/asterisk/sprachbausteine/#prompts-tts
>> These files are generated by our web-based text-to-speech engine.
>> Pros: If you need additional custom prompts, just go to
>> http://www.amooma.de/tts/ and generate them and the voice will
>> match.

> What is the license of these sound files generated on the website?

They are public domain (i.e. no license).


   Philipp Kempgen

-- 
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
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Re: [asterisk-users] What are the various models of DID providers

[SIP]

Excellent explanation, Alex.

What's interesting is the number of caveats and mixes even in the CLEC
and ILEC world.  I work with a CLEC that is also an ILEC (in certain
areas), since they encompass various areas in Georgia (and own the
state's largest contiguous network, passing through old rural ILEC lines
(now purchased and updated)). They maintain CLEC status in some areas
because they're not the incumbent there, but it helps them continue
their network across lines owned by the incumbent with various peering
agreements and the like.

One of the interesting things we ran across was a discussion with them
about UNEs. They provide strictly data lines throughout the state, and
their CLEC status allows them the purchase of UNE DS1s and DS3s at
exceptional rates to provide data to small installations in counties and
municipalities. However, upon reading the current governmental
regulations (the somewhat more recent E911 provisions), it states
specifically that a UNE MUST have, to each logical circuit, an assigned
DID and the ability to pass voice traffic to the local E911 call center.

The problem being, of course, that these were for data and not voice.
However, the law is very clear (in that murky way in which laws are),
and to avoid possible hassle down the road from an unfriendly ILEC or an
upset AT&T who wanted to press the issue, it was decided that DIDs would
be purchased and assigned to those UNE circuits as they were deployed.

This is where we came in, and where the middle-man model still works to
some degree. They could simply buy great swaths of DIDs for themselves
at ridiculously low rates (being a LEC), but the caveat there is that
the DIDs have to be USED, or they're reassigned. We stepped in to
provide DIDs (which we purchase elsewhere) to their UNE circuits and
maintain them (even with no use), as well as maintaining the information
for E911 dispatch on each of the circuits (assuming, for the sake of
argument, that someone were to convert the data line into voice). Thus,
they can get the rates they want on the UNEs they deserve, and not worry
about the hassles of actually dealing with the technology and contracts
on the voice side that is simply not part of their core business model.

Of course, their core model could change someday. But that would likely
involve a number of their personnel being devoted to the voice side of
things, as well as additional switching hardware, contract maintenance,
billing hassles, etc. All in all, we provide the services for a meager
fee (since we have the infrastructure  and personnel already in place),
and it's more cost-effective to go with a solution like that (and likely
will be for some time).

Now this is, to be certain, an odd and unusual case. I doubt we could
find too many customers if that were our ONLY sort of business. But it
does illustrate your point that there is still, for now, a logical place
for the middle men companies in some situations.

N.

Alex Balashov wrote:
> Hi Randulo,
>
> I think this topic is probably more appropriate for asterisk-biz, as was 
> the aforementioned rant about one particular DID provider.  But, 
> whatever - it is what it is.
>
> I assume that by "DID providers" you are referring to "origination" - 
> that is, picking up calls on PSTN numbers and converting them to VoIP 
> media and signaling and sending them to someone who wants to get numbers 
> that ordinary PSTN users can call on a VoIP system of some kind.  The 
> reason for the disambiguation is that many "DID providers" also provide 
> "termination" - that is, the delivery of calls from VoIP into the PSTN. 
>   There are also many companies that specialise in only origination or 
> termination.  The two are closely related from a technical perspective 
> but are characterised by rather different economics.
>
> At the end of the day--on a technical and a regulatory level--telephone 
> numbers can only belong to a carrier.  A carrier is a network operator 
> that is interconnected with other carriers and operates some form of 
> switch, and usually interfaces via SS7 (or CSS7, as it is known outside 
> North America) to the other carriers that they connect to.
>
> (Aside/digression about carriers:
>
> Of course, there are different types of carriers, depending on the 
> jurisdiction.
>
> In the US, there are - broadly speaking - two different types: 
> "incumbents" and "competitive" carriers involved in local service. 
> Incumbents are either Bell system entities that were divested from the 
> former AT&T monopoly in 1984 when AT&T was ordered to break itself up by 
> the federal government, or various local-yokel independent telephone 
> companies that were never acquired by AT&T during the 20th century (as 
> well as various types of conglomerates that have bought some of these 
> independents before, or since divestiture).  The latter type of 
> incumbent is usually in small towns and/or rural areas, whereas the 
> former is prevalent in metropolitan areas

Re: [asterisk-users] CDR Rewrite -- Questions to the users

[Benny Amorsen]

Steve Murphy  writes:

> Which of the two would you see being useful to you?

"Leg based", as far as I can see, because that looks like the only way
to bill transfers differently depending on which end did the transfer.

Possibly "Simple" on the Asterisk systems where we forbid transfers.

> Is there Yet Another CDR system you would like to see instead?
> How would/should it work?

"Leg based" looks good.

> Will both fulfil the requirements of CALEA?

We're not yet operating in a jurisdiction where CALEA applies. It
looks good enough for the jurisdictions we operate in, possibly apart
from the transfer issues further down, but I am certainly not a
lawyer.

> It's been proposed that we implement just the Simple 
> CDR now, and it be introduced in some 1.6.x (or higher)
> release.  In that release, the existing CDR system would be
> deprecated, and in some "futurer" release the "old" (now current)
> CDR system would be dropped entirely. What do you
> think? Are we high on drugs, or what?

I need this functionality for transfers, and I don't think "Simple"
provides it:

A calls B: A pays for the whole duration for A => B
B transfers to C: B pays for B => C, A is still paying A => B

If it was A who transferred the call instead:

A calls B: A pays for the whole duration for A => B
A transfers to C: A pays for A => C, and A is still paying A => B
B and C get to talk for free, while A pays twice.

This should apply whether transfers are attended (soft), unattended
(hard) or caused by SIP redirections before answering. Ideally it
should also be possible to simulate SIP-like redirections in the
dialplan with the same CDR behaviour.


/Benny


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[asterisk-users] Realtime MOH

[Max Alex]

Hi All,
I have set up realtime configuration of asterisk with mysql,
and it is working fine.
Asterisk version is :1.4.21
I have a issue regarding MOH, i have created musiconhold.conf in database as
per custom configuration.
When we reload moh then it is working fine, but some times the moh get
disappered and we must have to reload to load moh again.

Can any body please help me regarding MOH configuration!!

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Local channel Help required

[Max Alex]

Hi,
Thanks for your reply.
I have already used this
exten=> 1002,1,Dial(SIP/1002|30|rg)
exten=> 1002,2,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1002)

but my incoming call is getting hangup, it is not going to second priority.
So is there any configuration we have to do in local channel.

Thanks,
Max Alex
Voip Developer



On Mon, Jan 12, 2009 at 6:45 PM, Philipp Kempgen
wrote:

> Philipp Kempgen schrieb:
> > Max Alex schrieb:
> >
> >> If i got the NOANSWER then the channel is not passing to next priority.
> >> I need to pass that channel to the next priority of the context
> >> [macro-mypbx] so i can set voicemail there.
> >>
> >> I want to know how can we set the local channel to go in next priority
> in
> >> case of NO ANSWER.
> >
> > core show application Dial
> > ---cut---
> > g- Proceed with dialplan execution at the current extension if
> the
> >destination channel hangs up.
> > ---cut---
>
> No, wait, you don't need the g option here. Sorry.
>
> Dial() continues after ${DIALSTATUS} = NOANSWER anyway.
>
> Dial(SIP/${EXTEN});
> if ("${DIALSTATUS}" = "NOANSWER") {
>// go to voicemail
> }
>
>
>   Philipp Kempgen
>
> --
> AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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>
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[asterisk-users] Zaptel & multiple kernels

[Philipp Kempgen]

Hi,

If I have multiple kernel sources in /usr/src, e.g.
  linux-headers-2.6.26-1-686
  linux-headers-2.6.26.custom.1
how does the Zaptel Makefile(?) know which one to pick?

Is it a good approach to compile the kernel first and then compile
Zaptel "manually" afterwards?
Or should I rather put zaptel in /usr/src/modules and use
  fakeroot make-kpkg ... modules_image
in the kernel sources?


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Zaptel & multiple kernels

[Philipp Kempgen]

Philipp Kempgen schrieb:
> If I have multiple kernel sources in /usr/src, e.g.

headers

>   linux-headers-2.6.26-1-686
>   linux-headers-2.6.26.custom.1
> how does the Zaptel Makefile(?) know which one to pick?
> 
> Is it a good approach to compile the kernel first and then compile
> Zaptel "manually" afterwards?
> Or should I rather put zaptel in /usr/src/modules and use
>   fakeroot make-kpkg ... modules_image
> in the kernel sources?


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
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Re: [asterisk-users] Zaptel & multiple kernels

[Tzafrir Cohen]

On Tue, Jan 13, 2009 at 02:31:28PM +0100, Philipp Kempgen wrote:
> Hi,
> 
> If I have multiple kernel sources in /usr/src, e.g.
>   linux-headers-2.6.26-1-686
>   linux-headers-2.6.26.custom.1
> how does the Zaptel Makefile(?) know which one to pick?
> 
> Is it a good approach to compile the kernel first and then compile
> Zaptel "manually" afterwards?
> Or should I rather put zaptel in /usr/src/modules and use
>   fakeroot make-kpkg ... modules_image
> in the kernel sources?

By default:  /lib/modules/$KVERS/build

KVERS default to your kernel revision. e.g. `uname -r`, 2.6.26-1-686 .

This link will point to the appripriate "linux-headers" directory.

If you build your own kernel and install it using the kernel's 'install'
target, the 'build' link will point to your source tree.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] Dial() does not go to second priority (was: Re: Local channel Help required)

[Philipp Kempgen]

Max Alex schrieb:

> exten=> 1002,1,Dial(SIP/1002|30|rg)
> exten=> 1002,2,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1002)
> 
> but my incoming call is getting hangup, it is not going to second priority.

I hate ExecIf syntax but I don't see anything obvious here.
Could you try to send the DIALSTATUS to the CLI like so:

exten => 1002,1,Dial(SIP/1002|30|rg)
exten => 1002,n,Verbose(1,### DIALSTATUS: ${DIALSTATUS})
exten => 1002,n,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1002)


   Philipp Kempgen

-- 
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AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
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Re: [asterisk-users] Local channel Help required

[Doug Lytle]

Max Alex wrote:
> Hi,
> Thanks for your reply.
> I have already used this
> exten=> 1002,1,Dial(SIP/1002|30|rg)
> exten=> 1002,2,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1002)

This doesn't look correct (Based on looking at gotoif), try:


exten=> 1002,2,ExecIf($["${DIALSTATUS}" != "ANSWER" ]|Macro|voicedid|1002)

Doug

-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Zaptel & multiple kernels

[Philipp Kempgen]

Tzafrir Cohen schrieb:
> On Tue, Jan 13, 2009 at 02:31:28PM +0100, Philipp Kempgen wrote:
>> If I have multiple kernel sources in /usr/src, e.g.
>>   linux-headers-2.6.26-1-686
>>   linux-headers-2.6.26.custom.1
>> how does the Zaptel Makefile(?) know which one to pick?

> By default:  /lib/modules/$KVERS/build
> 
> KVERS default to your kernel revision. e.g. `uname -r`, 2.6.26-1-686 .

`uname -r` of the _running_ kernel that is? Good.

> This link will point to the appripriate "linux-headers" directory.
> 
> If you build your own kernel and install it using the kernel's 'install'
> target, the 'build' link will point to your source tree.

It does.
/lib/modules/2.6.26.custom.1/build -> /usr/src/linux-source-2.6.26

>> Is it a good approach to compile the kernel first and then compile
>> Zaptel "manually" afterwards?
>> Or should I rather put zaptel in /usr/src/modules and use
>>   fakeroot make-kpkg ... modules_image
>> in the kernel sources?

What about that?


   Philipp Kempgen

-- 
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
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[asterisk-users] 404 not found from one ip-adress

[Ralf Träskman]

Hi

Our sip provider has two servers that sends calls to our asterisk 1.6.
When server 1 sends call everything is working, but when server 2 sends call I 
get
[Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call 
from '' to extension '0840303390' rejected because extension not found.
And the provider get an "404 not found" error on their side.
What can be the problem?
Regards
/ralf

Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.com 
www.adlibris.com
P Please consider the environment before printing this e-mail

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Re: [asterisk-users] Zaptel & multiple kernels

[Philipp Kempgen]

Philipp Kempgen schrieb:
> Tzafrir Cohen schrieb:

>> By default:  /lib/modules/$KVERS/build
>> 
>> KVERS default to your kernel revision. e.g. `uname -r`, 2.6.26-1-686 .

>> This link will point to the appripriate "linux-headers" directory.
>> 
>> If you build your own kernel and install it using the kernel's 'install'
>> target, the 'build' link will point to your source tree.
> 
> It does.
> /lib/modules/2.6.26.custom.1/build -> /usr/src/linux-source-2.6.26

I guess it could be modified to point to
/usr/src/linux-headers-2.6.26.custom.1 instead which are the
kernel headers I generated with
fakeroot make-kpkg --initrd --append-to-version=.custom.1 kernel_headers


   Philipp Kempgen

-- 
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] What are the various models of DID providers

[Benny Amorsen]

Alex Balashov  writes:

> There are no exceptions to this rule;  numbers are assigned to carriers 
> and are switched and routed by carriers.  Where anyone is providing 
> DIDs, there is a UC (Underlying Carrier) involved that is actually doing 
> the hauling relative to the PSTN side.

Notice that in some areas it is now possible to become a carrier
without doing SS7. The company I work for is in the last stages of
transforming into a carrier, both for fixed and mobile. That is, we
now actually own our own fixed and mobile Danish number ranges and we
have our own SIM cards. We don't own any cell sites, but mobile calls
are actually switched through our infrastructure, not just handled by
the provider who owns the sites. This makes it possible to treat cell
phones as if they were SIP phones. As an example, Asterisk is able to
provide BLF showing the status of cell phones.

Interconnections with other carriers will be SIP only, and all of this
is done purely with Asterisk!

We only (ok, with a few exceptions) sell hosted PBX solutions, so we
aren't in general a price-competitive choice for pure
termination/origination.


/Benny

PS: Sorry if this seems like an advertisement, but I do believe that
what we are doing is a new and exciting direction for Asterisk. Like
in my other posts, I have avoided mentioning the company I work for.

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Re: [asterisk-users] 404 not found from one ip-adress

[Dovid Bender]

  - Original Message - 
  From: Ralf Träskman 
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  Sent: Tuesday, January 13, 2009 4:04 PM
  Subject: [asterisk-users] 404 not found from one ip-adress


  Hi

   

  Our sip provider has two servers that sends calls to our asterisk 1.6.

  When server 1 sends call everything is working, but when server 2 sends call 
I get 

  [Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call 
from '' to extension '0840303390' rejected because extension not found.

  And the provider get an "404 not found" error on their side.

  What can be the problem?

  Regards

  /ralf

  

  Ralf Träskman, IT
  AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
  Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
  r...@adlibris.com www.adlibris.com 
  P Please consider the environment before printing this e-mail

Raif,
What does your sip register statement look like ? It seems that they are 
sending it to yournum...@yourip and you do not have it set up in the context 
for this carrier. You can call them and ask them to fix it or just add in Exten 
=> 084303390 in the context and then just have a goto to the extension that the 
first server is sending calls to (maybe the s extension ?).


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Re: [asterisk-users] Zaptel & multiple kernels

[Tzafrir Cohen]

On Tue, Jan 13, 2009 at 03:03:04PM +0100, Philipp Kempgen wrote:
> Tzafrir Cohen schrieb:
> > On Tue, Jan 13, 2009 at 02:31:28PM +0100, Philipp Kempgen wrote:
> >> If I have multiple kernel sources in /usr/src, e.g.
> >>   linux-headers-2.6.26-1-686
> >>   linux-headers-2.6.26.custom.1
> >> how does the Zaptel Makefile(?) know which one to pick?
> 
> > By default:  /lib/modules/$KVERS/build
> > 
> > KVERS default to your kernel revision. e.g. `uname -r`, 2.6.26-1-686 .
> 
> `uname -r` of the _running_ kernel that is? Good.
> 
> > This link will point to the appripriate "linux-headers" directory.
> > 
> > If you build your own kernel and install it using the kernel's 'install'
> > target, the 'build' link will point to your source tree.
> 
> It does.
> /lib/modules/2.6.26.custom.1/build -> /usr/src/linux-source-2.6.26
> 
> >> Is it a good approach to compile the kernel first and then compile
> >> Zaptel "manually" afterwards?
> >> Or should I rather put zaptel in /usr/src/modules and use
> >>   fakeroot make-kpkg ... modules_image
> >> in the kernel sources?
> 
> What about that?

In the worst case, set KSRC explicitly.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] What are the various models of DID providers

[randulo]

On Tue, Jan 13, 2009 at 3:11 PM, Benny Amorsen  wrote:
> PS: Sorry if this seems like an advertisement, but I do believe that
> what we are doing is a new and exciting direction for Asterisk. Like
> in my other posts, I have avoided mentioning the company I work for.

Not at all, you added pertinent info. Since I'm getting  good info,
I've decided to make this the subject for the VoIP Users Conference
this Friday. You are cordially invited (Ben and everyone else) to join
the discussion to help define the spaces discussed here. That would be
at 6PM in DK if you are on the same time as Paris?

Email me directly if you need more info on how to call.

/r

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Re: [asterisk-users] 404 not found from one ip-adress

[Danny Nicholas]

Provider 2 is dropping into a new context than Provider 1.  The $EXTEN is
probably coming in from P1 as XX and P2 as AXX.  Check your incoming
and default sections of extensions.conf.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralf Träskman
Sent: Tuesday, January 13, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] 404 not found from one ip-adress

 

Hi

 

Our sip provider has two servers that sends calls to our asterisk 1.6.

When server 1 sends call everything is working, but when server 2 sends call
I get 

[Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite:
Call from '' to extension '0840303390' rejected because extension not found.

And the provider get an “404 not found” error on their side.

What can be the problem?

Regards

/ralf



Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.com www.adlibris.com   
P Please consider the environment before printing this e-mail

 

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Re: [asterisk-users] 404 not found from one ip-adress

[Ralf Träskman]

Hi

The provider dont use register, they are running openSER I have this in my 
sip.conf

[outgoing]
context=ip-only
disallow=all
allow=alaw,ulaw
canreinvite=yes
dtmfmode=rfc2833
host=sip.hub.ip-only.se
insecure=very
reinvite=yes
type=friend

[incoming]
disallow=all
allow=alaw,ulaw
context=ip-only
type=user

Regards
/ralf

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender
Sent: den 13 januari 2009 15:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 404 not found from one ip-adress



- Original Message -
From: Ralf Träskman
To: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'
Sent: Tuesday, January 13, 2009 4:04 PM
Subject: [asterisk-users] 404 not found from one ip-adress

Hi

Our sip provider has two servers that sends calls to our asterisk 1.6.
When server 1 sends call everything is working, but when server 2 sends call I 
get
[Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call 
from '' to extension '0840303390' rejected because extension not found.
And the provider get an "404 not found" error on their side.
What can be the problem?
Regards
/ralf

Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.com 
www.adlibris.com
P Please consider the environment before printing this e-mail
Raif,
What does your sip register statement look like ? It seems that they are 
sending it to yournum...@yourip and you do not have 
it set up in the context for this carrier. You can call them and ask them to 
fix it or just add in Exten => 084303390 in the context and then just have a 
goto to the extension that the first server is sending calls to (maybe the s 
extension ?).



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Re: [asterisk-users] 404 not found from one ip-adress

[Ralf Träskman]

Hi

Its the same provider and i use dns name in sip.conf

[outgoing]
context=ip-only
disallow=all
allow=alaw,ulaw
canreinvite=yes
dtmfmode=rfc2833
host=sip.hub.ip-only.se
insecure=very
reinvite=yes
type=friend

[incoming]
disallow=all
allow=alaw,ulaw
context=ip-only
type=user

Regards
/ralf

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: den 13 januari 2009 15:39
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 404 not found from one ip-adress

Provider 2 is dropping into a new context than Provider 1.  The $EXTEN is 
probably coming in from P1 as XX and P2 as AXX.  Check your incoming 
and default sections of extensions.conf.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralf Träskman
Sent: Tuesday, January 13, 2009 8:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] 404 not found from one ip-adress

Hi

Our sip provider has two servers that sends calls to our asterisk 1.6.
When server 1 sends call everything is working, but when server 2 sends call I 
get
[Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call 
from '' to extension '0840303390' rejected because extension not found.
And the provider get an "404 not found" error on their side.
What can be the problem?
Regards
/ralf

Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.com 
www.adlibris.com
P Please consider the environment before printing this e-mail

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Re: [asterisk-users] 404 not found from one ip-adress

[Kristian Kielhofner]

On Tue, Jan 13, 2009 at 9:59 AM, Ralf Träskman  wrote:
> Hi
>
>
>
> The provider dont use register, they are running openSER I have this in my
> sip.conf
>
>
>
> [outgoing]
>
> context=ip-only
>
> disallow=all
>
> allow=alaw,ulaw
>
> canreinvite=yes
>
> dtmfmode=rfc2833
>
> host=sip.hub.ip-only.se
>
> insecure=very
>
> reinvite=yes
>
> type=friend
>
>
>
> [incoming]
>
> disallow=all
>
> allow=alaw,ulaw
>
> context=ip-only
>
> type=user
>
>
>
> Regards
>
> /ralf
>

Ralf,

  That incoming peer isn't matching anything.

  They're probably hitting the context defined in [general].  Add
another peer/friend match with the other servers IP/hostname and the
ip-only context.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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[asterisk-users] [Re: CDR Rewrite -- Questions to the users]

[Steve Murphy]

Benny--

Thanks for the response!  I've inserted comments in the following:

PS. Pardon the HTML format; my email editor splits lines at an
unadjustably
small number of columns, but in HTML, no line length limits, and better
looking examples!

On Tue, 2009-01-13 at 14:16 +0100, Benny Amorsen wrote: 

> Steve Murphy  writes:
> 
> > Which of the two would you see being useful to you?
> 
> "Leg based", as far as I can see, because that looks like the only way
> to bill transfers differently depending on which end did the transfer.
> 
> Possibly "Simple" on the Asterisk systems where we forbid transfers.
> 
> > Is there Yet Another CDR system you would like to see instead?
> > How would/should it work?
> 
> "Leg based" looks good.
> 
> > Will both fulfil the requirements of CALEA?
> 
> We're not yet operating in a jurisdiction where CALEA applies. It
> looks good enough for the jurisdictions we operate in, possibly apart
> from the transfer issues further down, but I am certainly not a
> lawyer.
> 
> > It's been proposed that we implement just the Simple 
> > CDR now, and it be introduced in some 1.6.x (or higher)
> > release.  In that release, the existing CDR system would be
> > deprecated, and in some "futurer" release the "old" (now current)
> > CDR system would be dropped entirely. What do you
> > think? Are we high on drugs, or what?
> 
> I need this functionality for transfers, and I don't think "Simple"
> provides it:
> 
> A calls B: A pays for the whole duration for A => B
> B transfers to C: B pays for B => C, A is still paying A => B
> 


Good Question: Can Simple CDRs be used in xfer situations?

Let's take a look.

In this particular situation, 3 channels are involved: A, B, and C.
Therefore,
you will get 3 CDRs.

 CDR1:  A -> B  start: e1a  ans: e2  end: e4   Party: B  disp: ANSW
linkedID: abc9
 CDR2:  A   start: e1   ans: e1  end: e6   Party: A  disp: ANSW
linkedID: abc9
 CDR3:  B -> C  start: e4   ans: e5  end: e6   Party: C  disp: ANSW
linkedid: abc9

CDR2 covers A (see the Party field),  CDR1 covers B, CDR3 covers C.

A's CDR could be used to bill A for his call in. It covers both the time
A spent
talking to B, and C. If you charge a different rate for A talking to B
vs C, then
you have some interesting SQL queries to make, I'll guess...

C's CDR records that B called C. It doesn't mention that A is doing all
the talking.

B's CDR records the call from A to B; this is the only one that seems a
little useless...

Is this enough? If this is all you had, could you make it work? If you
can't, 
would adding a field or two help?



> If it was A who transferred the call instead:
> 
> A calls B: A pays for the whole duration for A => B
> A transfers to C: A pays for A => C, and A is still paying A => B
> B and C get to talk for free, while A pays twice.
> 


In the SImple CDR world, here's what would be produced:

 CDR1:  A   start: e1   ans: e1  end: e4   Party: A  disp: ANSW
linkedID: abc9
 CDR2:  A -> B  start: e1a  ans: e2  end: e6   Party: B  disp: ANSW
linkedID: abc9
 CDR3:  A -> C  start: e4   ans: e5  end: e6   Party: C  disp: ANSW
linkedid: abc9

Here, A's total connection time is in CDR1; B with CDR2;  C with CDR3.

The call from B to C is in CDR3. A's transfer to C is in CDR3 (I just
corrected this
in the CDRfix2 document).

Again, is there enough info here for you to do what you need to do? If
not
what addition could be added to make it work?



> This should apply whether transfers are attended (soft), unattended
> (hard) or caused by SIP redirections before answering. Ideally it
> should also be possible to simulate SIP-like redirections in the
> dialplan with the same CDR behaviour.
> 


In the CDRfix2 doc, I outlined both the above blindxfer cases, and also
permutations
of attended xfers. Look them over, and see if what you need is possible
with this format,
and if not, is there something we can add that *would* make it
usable...?

murf

-- 
Steve Murphy 
Digium



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Re: [asterisk-users] Zaptel & multiple kernels

[Philipp Kempgen]

Tzafrir Cohen schrieb:
> On Tue, Jan 13, 2009 at 03:03:04PM +0100, Philipp Kempgen wrote:
>> Tzafrir Cohen schrieb:
>> > On Tue, Jan 13, 2009 at 02:31:28PM +0100, Philipp Kempgen wrote:

>> >> Is it a good approach to compile the kernel first and then compile
>> >> Zaptel "manually" afterwards?
>> >> Or should I rather put zaptel in /usr/src/modules and use
>> >>   fakeroot make-kpkg ... modules_image
>> >> in the kernel sources?

> In the worst case, set KSRC explicitly.

Er. Never mind.

Thanks for your explanation!


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] Upgrade Cisco 7971G-GE from SCCP to SIP

[Ayman Boules (Live.COM)]

Good Morning Everyone,

It will be great if someone can help me upgrade a Cisco 7971G-GE to SIP.  If
so, please email me the detailed instructions to do the upgrade.

I will appreciate it much if you have the latest 8.4(2) firmware (file name:
cmterm-7970_7971-sip.8-4-2.cop) and email it to me or send me a link to
download it...



Regards,

Ayman L. Boules
Sunday, January 11, 2009
++

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason
(Lists)
Sent: Sunday, January 11, 2009 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] chan_sip on non-standard port 5062 - contact
has no port

You are configuring Asterisk to LISTEN on 5062 , if you want it to talk 
to another server on 5062, then configure that server's config stanza 
accordingly.

-- 
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[asterisk-users] 0800 UK number

[Julian Lyndon-Smith]

I have concocted a system for my children's primary school where parents 
can dial in and subscribe to an "emergency broadcast" message so that 
they can be automatically contacted in case of a problem like the school 
being shut because of snow etc.

I would like to provide an 0800 number service for this, so that there 
is no cost to the parents, but obviously I would like to get the best 
package possible.

I have come across several packages, but would like the "most inclusive 
minutes" for "the best price" ;)

Does anyone that has used an 0800 service in the UK have any 
recomendations ?

Thanks

Julian.


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[asterisk-users] mISDN BRI Asterisk 1.4

[Lee Wilson]

Good Evening,

The company I work for is attempting to connect an Cisco ISDN Router to an 
OpenVOX B200P BRI Card so that we can get it to dial out across an existing 
ISDN PRI Line also installed in the Asterisk PBX

Everything has compiled successfully and the BRI card has been detected when 
mISDN is started:

# misdnportinfo
Port  1: NT-mode BRI S/T interface port (for phones)
 -> Interface can be Poin-To-Point/Multipoint.

Port  2: NT-mode BRI S/T interface port (for phones)
 -> Interface can be Poin-To-Point/Multipoint.


However when checking in Asterisk Layer 1 is always down even though the Router 
is connected:

*CLI> misdn show stacks
BEGIN STACK_LIST:
  * Port 1 Type NT Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0  Debug:0
  * Port 2 Type NT Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0  Debug:0

I've tried configuring the Ports on the BRI card to be in both TE and NT Mode 
(as shown above) as well as using link=ptp or ptmp in the mISDN.conf file:


hfcmulti
mISDN_dsp
mISDN

1
2
3
4



When plugging the Cisco router into our existing Nortel switch the Link light 
comes up immediately. But on the new Asterisk PBX the BRI card is constantly 
flashing Red with no lights on either Channel 1 or 2 on the Router.

I'd appreciate any help that can be offered.

Thanks

Lee


  


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Re: [asterisk-users] 0800 UK number

[David fire]

why 0800? the parents will subscribe to the system only once
you have a  lot of flat fee services on-line to call land lines/mobiles in
UK.
David


2009/1/13 Julian Lyndon-Smith 

> I have concocted a system for my children's primary school where parents
> can dial in and subscribe to an "emergency broadcast" message so that
> they can be automatically contacted in case of a problem like the school
> being shut because of snow etc.
>
> I would like to provide an 0800 number service for this, so that there
> is no cost to the parents, but obviously I would like to get the best
> package possible.
>
> I have come across several packages, but would like the "most inclusive
> minutes" for "the best price" ;)
>
> Does anyone that has used an 0800 service in the UK have any
> recomendations ?
>
> Thanks
>
> Julian.
>
>
> __
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Re: [asterisk-users] 0800 UK number

[Julian Lyndon-Smith]

The number will also be used as a "information line" where a more 
detailed message can be played.

The scenario is:

1) School is closed because the boiler has broken down.
2) The Head (or any authorised person) calls the service and leaves a 
detailed message of the reason for closure
3) The system sends the text message* to all subscribers saying "School 
is closed today, please call foo for more details"
4) If anyone wants more details, then they call in.

* this message can be changed at any time by the Head texting it to the 
service.

Julian

David fire wrote:
> why 0800? the parents will subscribe to the system only once
> you have a  lot of flat fee services on-line to call land 
> lines/mobiles in UK.
> David
>
>
> 2009/1/13 Julian Lyndon-Smith  >
>
> I have concocted a system for my children's primary school where
> parents
> can dial in and subscribe to an "emergency broadcast" message so that
> they can be automatically contacted in case of a problem like the
> school
> being shut because of snow etc.
>
> I would like to provide an 0800 number service for this, so that there
> is no cost to the parents, but obviously I would like to get the best
> package possible.
>
> I have come across several packages, but would like the "most
> inclusive
> minutes" for "the best price" ;)
>
> Does anyone that has used an 0800 service in the UK have any
> recomendations ?
>
> Thanks
>
> Julian.
>
>
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>
>
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Re: [asterisk-users] mISDN BRI Asterisk 1.4

[Olivier]

2009/1/13 Lee Wilson 

> Good Evening,
>
> The company I work for is attempting to connect an Cisco ISDN Router to an
> OpenVOX B200P BRI Card so that we can get it to dial out across an existing
> ISDN PRI Line also installed in the Asterisk PBX
>

Hello,

Is your setup like this ?
ISDN ---  Asterisk box with PRI and BRI boards  Cisco
router whatever


Cheers
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[asterisk-users] Problem with overhead paging with Alsa and OSS

[Carlos Chavez]

I recently upgraded a server to Asterisk 1.4.22 with OpenR2.
Previously I was using 1.4.18.  It seems that 1.4.22 has a big bug using
chan_alsa.so for overhead paging.  After rebooting the server it would
work once or twice and then I just got an error on the CLI:

[Jan  7 10:35:14] ERROR[26164]: chan_alsa.c:693 alsa_read: Read error:
Resource temporarily unavailable

I had to switch to chan_oss using the oss emulation in alsa but now the
customer complains that the voice is metallic sounding and intermittent.
What could cause this problem with the voice?  With 1.4.18 and chan_alsa
you could page from any phone (Zap and SIP) and the voice was always
clear.


-- 
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Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Upgrade Cisco 7971G-GE from SCCP to SIP

[Steve Edwards]

On Tue, 13 Jan 2009, Ayman Boules (Live.COM) wrote:

> It will be great if someone can help me upgrade a Cisco 7971G-GE to SIP. 
> If so, please email me the detailed instructions to do the upgrade.

Where's that link to "http://letmegogglethatforyou.com?";

> I will appreciate it much if you have the latest 8.4(2) firmware (file 
> name: cmterm-7970_7971-sip.8-4-2.cop) and email it to me or send me a 
> link to download it...

Oh. Of course. Let's all violate cisco's copyright on a public mailing 
list :)

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] 0800 UK number

[Gordon Henderson]

On Tue, 13 Jan 2009, Julian Lyndon-Smith wrote:

> I have concocted a system for my children's primary school where parents
> can dial in and subscribe to an "emergency broadcast" message so that
> they can be automatically contacted in case of a problem like the school
> being shut because of snow etc.
>
> I would like to provide an 0800 number service for this, so that there
> is no cost to the parents, but obviously I would like to get the best
> package possible.
>
> I have come across several packages, but would like the "most inclusive
> minutes" for "the best price" ;)
>
> Does anyone that has used an 0800 service in the UK have any
> recomendations ?

Are you looking for VoIP presentation (SIP/IAX), or plumbed to an 01/02 
number?

Remember that 0800's may not be free to call from mobiles, but 01/02/03 
numbers might come out of their inclusive minutes, and I'd guess more 
people are using inclusive call packages from their telcos now too, so I'm 
wondering if 0800's really are the best thing these days...

Gordon

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Re: [asterisk-users] CDR Rewrite -- Questions to the users

[Benny Amorsen]

I wrote a really long email, but it hinged on one thing I need
clarified...

tir, 13 01 2009 kl. 09:05 -0700, skrev Steve Murphy:

>  CDR1:  A -> B  start: e1a  ans: e2  end: e4   Party: B  disp:
> ANSW   linkedID: abc9
>  CDR2:  A   start: e1   ans: e1  end: e6   Party: A  disp:
> ANSW   linkedID: abc9


We are talking about the "Simple" CDR's, not the leg-based ones, right?
If so, why do all the CDR's only call one "Party"? Shouldn't there be a
src and a destination? 


/Benny


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Re: [asterisk-users] 0800 UK number

[Julian Lyndon-Smith]

Hi Gordon,

Gordon Henderson wrote:
> On Tue, 13 Jan 2009, Julian Lyndon-Smith wrote:
>
>   
>> I have concocted a system for my children's primary school where parents
>> can dial in and subscribe to an "emergency broadcast" message so that
>> they can be automatically contacted in case of a problem like the school
>> being shut because of snow etc.
>>
>> I would like to provide an 0800 number service for this, so that there
>> is no cost to the parents, but obviously I would like to get the best
>> package possible.
>>
>> I have come across several packages, but would like the "most inclusive
>> minutes" for "the best price" ;)
>>
>> Does anyone that has used an 0800 service in the UK have any
>> recomendations ?
>> 
>
> Are you looking for VoIP presentation (SIP/IAX), or plumbed to an 01/02 
> number?
>
> Remember that 0800's may not be free to call from mobiles, but 01/02/03 
> numbers might come out of their inclusive minutes, and I'd guess more 
>   
That's a fair point.
> people are using inclusive call packages from their telcos now too, so I'm 
> wondering if 0800's really are the best thing these days...
>   
The only "fly" in the ointment is that my server is in 01702, but I need 
a "local number" (01376) for "political" reasons

Julian
> Gordon
>
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>   


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Re: [asterisk-users] mISDN BRI Asterisk 1.4

[Lee Wilson]

On Tue, Jan 13, 2009 at 6:35 PM, Olivier wrote:
>
>
> 2009/1/13 Lee Wilson
>>
>> Good Evening,
>>
>> The company I work for is attempting to connect an Cisco ISDN Router to an 
>> OpenVOX B200P BRI Card so that we can get it to dial out across an existing 
>> ISDN PRI Line also installed in the Asterisk PBX
>
> Hello,
>
> Is your setup like this ?
> ISDN ---  Asterisk box with PRI and BRI boards  Cisco 
> router whatever
>
>
> Cheers

Hi Olivier,

That is spot on.  I'm not sure if we have 2-4 PRI Channels dedicated to the BRI 
or if they are allocated dynamically but don't think it matter anyway.

Thanks

Lee


  


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[asterisk-users] test

[David @ULC]

test
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[asterisk-users] FWD and Asterisk

[David @ULC]

I have an account with FWD and I have configured my SIP.conf with

[fwd]
type=friend
secret=password
username=901835
host=fwd.pulver.com

But when I am trying to dial out my own DID , I dont see any call landing in
asterisk.

In extension.conf (vicidial) file I have

exten => 2062036895 ,1,Ringing()
exten => 2062036895 ,2,Wait(1)
exten => 2062036895 ,3,Answer()
exten => 2062036895
,4,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-2062036895-Closer---999-1)
exten => 2062036895 ,5,Hangup()

2062036895 is with IPKall.

What could be the wrong. ?
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[asterisk-users] FWD and Asterisk

[David @ULC]

I also tried but cant see any call landing up in asterisk.

Btw, how to find out whether a call is landing in Asterisk or not ?

[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very<<< this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
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Re: [asterisk-users] 0800 UK number

[Thomas Kenyon]

Julian Lyndon-Smith wrote:

> The only "fly" in the ointment is that my server is in 01702, but I need 
> a "local number" (01376) for "political" reasons
> 
That's hardly a problem, (If the call is to be presented using VoIP) 
more or less any provider will give you a local number from another area.

I have an 0800 number with voiptalk (an ex-BT number) and that seems to 
be pretty reliable.

Although they did put the price up a year ago without telling me.

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Re: [asterisk-users] 0800 UK number

[Julian Lyndon-Smith]

Thomas Kenyon wrote:
> Julian Lyndon-Smith wrote:
>
>> The only "fly" in the ointment is that my server is in 01702, but I 
>> need a "local number" (01376) for "political" reasons
>>
> That's hardly a problem, (If the call is to be presented using VoIP) 
> more or less any provider will give you a local number from another area.
>
Unfortunately, it's not to be presented by VOIP - the server uses isdn30.
> I have an 0800 number with voiptalk (an ex-BT number) and that seems 
> to be pretty reliable.
>
> Although they did put the price up a year ago without telling me.
>

Julian

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Re: [asterisk-users] FWD and Asterisk

[Jai Rangi]

ngrep port 5060
or tcpdum port 5060
By default asterisk runs on port 5060, that way you can see if your getting
the signal or not.

Jai Rangi
"Buy SIP DID www.didforsale.com
free Trial now purchase required"

On Tue, Jan 13, 2009 at 1:13 PM, David @ULC  wrote:

> I also tried but cant see any call landing up in asterisk.
>
> Btw, how to find out whether a call is landing in Asterisk or not ?
>
> [123]
> type=peer
> qualify=no
> port=5060
> nat=no
> insecure=very<<< this is very important
> host=voiper.ipkall.com
> dtmfmode=rfc2833
> context=from-pstn
> canreinvite=no
>
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[asterisk-users] FWD and Asterisk

[David @ULC]

[r...@vicidialnow ~]# ngrep port 5060
-bash: ngrep: command not found
[r...@vicidialnow ~]# tcpdum port 5060
-bash: tcpdum: command not found
[r...@vicidialnow ~]#


Also, is my SIP configuration is correct ?
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Re: [asterisk-users] FWD and Asterisk

[Jai Rangi]

Sorry for the typo,
tcpdump port 5060
ngrep you can download the rpm (google) easy to install
http://rpm.pbone.net/index.php3/stat/4/idpl/1127130/com/ngrep-1.38-1.i386.rpm.html
rpm -ivh 
ngrep-1.38-1.i386.rpm

Is you sip configuration right?
cant tell without looking at it.

Jai Rangi
"Buy SIP DID www.didforsale.com
free Trial no purchase required"


On Tue, Jan 13, 2009 at 1:44 PM, David @ULC  wrote:

>
> [r...@vicidialnow ~]# ngrep port 5060
> -bash: ngrep: command not found
> [r...@vicidialnow ~]# tcpdum port 5060
> -bash: tcpdum: command not found
> [r...@vicidialnow ~]#
>
>
> Also, is my SIP configuration is correct ?
>
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[asterisk-users] FWD and Asterisk

[David @ULC]

[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very<<< this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
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Re: [asterisk-users] FWD and Asterisk

[Philipp Kempgen]

David @ULC schrieb:
> [r...@vicidialnow ~]# ngrep port 5060
> -bash: ngrep: command not found

aptitude install ngrep

> [r...@vicidialnow ~]# tcpdum port 5060
> -bash: tcpdum: command not found

aptitude install tcpdump


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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[asterisk-users] FWD and Asterisk

[David @ULC]

Anyone using FWD with Asterisk ?

On Wed, Jan 14, 2009 at 2:40 AM, David @ULC  wrote:

>
> I have an account with FWD and I have configured my SIP.conf with
>
> [fwd]
> type=friend
> secret=password
> username=901835
> host=fwd.pulver.com
>
> But when I am trying to dial out my own DID , I dont see any call landing
> in asterisk.
>
> In extension.conf (vicidial) file I have
>
> exten => 2062036895 ,1,Ringing()
> exten => 2062036895 ,2,Wait(1)
> exten => 2062036895 ,3,Answer()
> exten => 2062036895
> ,4,AGI(agi-VDADcloser_inboundCIDlookup.agi,SALESLINE-2062036895-Closer---999-1)
> exten => 2062036895 ,5,Hangup()
>
> 2062036895 is with IPKall.
>
> What could be the wrong. ?
>
>
>
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[asterisk-users] Asterisk Appliance

[Bob Pierce]

I'm looking for some info on the Asterisk Appliance.

I understand it has a gui, but can I still do all the dialplan config
that I'm used of doing by hand outside of the gui? If I really wanted
to, could I even ignore that the device has a gui and do all my config
in the files? I guess I'm just wondering if it will be as flexible as a
'vanilla' asterisk install from source on a linux system.

Also, from those who are using these devices, what has your experience
been? Are they stable? Do they seem to have enough horsepower and
storage space for an SMB with up to 50 phones? Some older specs stated
they would be appropriate for businesses with 2-50 users, while the
current spec on the Digium site states they are appropriate for 2-20
users.

The application I'm thinking of would be VoIP only with a g.711 SIP
trunk and g.711 phones.

Thanks for your input.

Bob

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[asterisk-users] FWD and IPCall

[David @ULC]

I tried this
http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html

But I am NOT getting call in asterisk.


SIP.conf file :
_

[general]
port = 5060
bindaddr = 0.0.0.0
context = default
externhost=59.160.44.21
localnet=192.168.0.2/255.255.255.0
; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:t...@10.10.10.16:5060
;

[sip]
type=peer
username=fiducia_ag
fromuser=fiducia_ag
authuser=fiducia_ag
secret=password
host=64.56.64.64
nat=no
canreinvite=yes
insecure=very
disallow=all
allow=g729
allow=ulaw
context=default
dtmfmode=rfc2833

[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw

Extension.conf:
_

[from-ipkall]
exten => 901835,1,NoOp(from-ipkall)
exten => 901835,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten => 901835,3,Dial(Local/200 at internal)
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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

SIP wrote:

> What's interesting is the number of caveats and mixes even in the CLEC
> and ILEC world.  I work with a CLEC that is also an ILEC (in certain
> areas), since they encompass various areas in Georgia (and own the
> state's largest contiguous network, passing through old rural ILEC lines
> (now purchased and updated)). They maintain CLEC status in some areas
> because they're not the incumbent there, but it helps them continue
> their network across lines owned by the incumbent with various peering
> agreements and the like.

GTA?

> One of the interesting things we ran across was a discussion with them
> about UNEs. They provide strictly data lines throughout the state, and
> their CLEC status allows them the purchase of UNE DS1s and DS3s at
> exceptional rates to provide data to small installations in counties and
> municipalities. 

I don't know that the price of UNE DS1s and DS3s is really all that 
exceptional.  Sure, it seems impressive that you can get a T1 in LATA 
438 for some odd $44, but once you factor in the costs of 
interconnection, CO colocation, EELs and interoffice mileage if not 
colocated in the CO to which the circuit is being generated, private 
SONET for backhaul, etc.

Not to mention in that in urban areas the ILEC commonly suspends UNE 
pricing discipline on the grounds that the wire center is "impaired" - 
i.e. there is enough "competition" in the CO.  That requires you to 
revert to wholesale / special access and pay a lot more.

> However, upon reading the current governmental
> regulations (the somewhat more recent E911 provisions), it states
> specifically that a UNE MUST have, to each logical circuit, an assigned
> DID and the ability to pass voice traffic to the local E911 call center.
> 
> The problem being, of course, that these were for data and not voice.
> However, the law is very clear (in that murky way in which laws are),
> and to avoid possible hassle down the road from an unfriendly ILEC or an
> upset AT&T who wanted to press the issue, it was decided that DIDs would
> be purchased and assigned to those UNE circuits as they were deployed.

I'm not sure I follow.  Voice trunks need routing to E911 tandems, but 
what do data circuits have to do with this?

> This is where we came in, and where the middle-man model still works to
> some degree. They could simply buy great swaths of DIDs for themselves
> at ridiculously low rates (being a LEC), but the caveat there is that
> the DIDs have to be USED, or they're reassigned. 

Depends on the area;  NANPA and pooling blocks aren't necessarily cheap.

> We stepped in to
> provide DIDs (which we purchase elsewhere) to their UNE circuits and
> maintain them (even with no use), as well as maintaining the information
> for E911 dispatch on each of the circuits (assuming, for the sake of
> argument, that someone were to convert the data line into voice). Thus,
> they can get the rates they want on the UNEs they deserve, and not worry
> about the hassles of actually dealing with the technology and contracts
> on the voice side that is simply not part of their core business model.

Why would they have to deal with this when someone buying directly from 
AT&T off the special access tariff doesn't?  (i.e. independent ISPs)

> Now this is, to be certain, an odd and unusual case. I doubt we could
> find too many customers if that were our ONLY sort of business. But it
> does illustrate your point that there is still, for now, a logical place
> for the middle men companies in some situations.

Agreed, although I'm still very confused as to why you need DIDs for 
data UNEs.  Is this some bizarre feature of their ICA or something?

-- Alex

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

Benny Amorsen wrote:
> Alex Balashov  writes:
> 
>> There are no exceptions to this rule;  numbers are assigned to carriers 
>> and are switched and routed by carriers.  Where anyone is providing 
>> DIDs, there is a UC (Underlying Carrier) involved that is actually doing 
>> the hauling relative to the PSTN side.
> 
> Notice that in some areas it is now possible to become a carrier
> without doing SS7. The company I work for is in the last stages of
> transforming into a carrier, both for fixed and mobile. That is, we
> now actually own our own fixed and mobile Danish number ranges and we
> have our own SIM cards. We don't own any cell sites, but mobile calls
> are actually switched through our infrastructure, not just handled by
> the provider who owns the sites. This makes it possible to treat cell
> phones as if they were SIP phones. As an example, Asterisk is able to
> provide BLF showing the status of cell phones.
> 
> Interconnections with other carriers will be SIP only, and all of this
> is done purely with Asterisk!

That's a highly progressive jurisdiction to be in, then.  :-)  In much 
of the rest of the world, you have to be doing SS7.  You can't connect 
to any ILECs in the US any other way.

That aside, it is replace internal private SS7 IMTs between switches (if 
you have multiple ones) with SIP-T and avoid the limitations of physical 
trunk exhaust and the need for separate networks.

Also, there are ways to avoid having to take SS7 A-links directly even 
if you have point codes.  You can get somebody to do it for you and give 
you SIGTRAN over IP - even over the Internet if you must.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] mISDN BRI Asterisk 1.4

[Ex Vito]

  While I don't know the OpenVOX B200P specifics, some interface cards
  need you to change physical jumpers in order to acheive NT vs TE, mode.

  Could that be the case ?
--
  exvito

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Re: [asterisk-users] What are the various models of DID providers

[SIP]

Alex Balashov wrote:
> SIP wrote:
>
>   
>> What's interesting is the number of caveats and mixes even in the CLEC
>> and ILEC world.  I work with a CLEC that is also an ILEC (in certain
>> areas), since they encompass various areas in Georgia (and own the
>> state's largest contiguous network, passing through old rural ILEC lines
>> (now purchased and updated)). They maintain CLEC status in some areas
>> because they're not the incumbent there, but it helps them continue
>> their network across lines owned by the incumbent with various peering
>> agreements and the like.
>> 
>   
>> One of the interesting things we ran across was a discussion with them
>> about UNEs. They provide strictly data lines throughout the state, and
>> their CLEC status allows them the purchase of UNE DS1s and DS3s at
>> exceptional rates to provide data to small installations in counties and
>> municipalities. 
>> 
>
> I don't know that the price of UNE DS1s and DS3s is really all that 
> exceptional.  Sure, it seems impressive that you can get a T1 in LATA 
> 438 for some odd $44, but once you factor in the costs of 
> interconnection, CO colocation, EELs and interoffice mileage if not 
> colocated in the CO to which the circuit is being generated, private 
> SONET for backhaul, etc.
>
> Not to mention in that in urban areas the ILEC commonly suspends UNE 
> pricing discipline on the grounds that the wire center is "impaired" - 
> i.e. there is enough "competition" in the CO.  That requires you to 
> revert to wholesale / special access and pay a lot more.
>   

The interconnection, CO colocation, private SONET, etc, are already in
place in something like 60 municipalities and 4 Atlanta metro areas.
They're using the UNEs to cut costs. Honestly, you could ask me some
complex questions about their network, but I don't know it all that well...


>   
>> However, upon reading the current governmental
>> regulations (the somewhat more recent E911 provisions), it states
>> specifically that a UNE MUST have, to each logical circuit, an assigned
>> DID and the ability to pass voice traffic to the local E911 call center.
>>
>> The problem being, of course, that these were for data and not voice.
>> However, the law is very clear (in that murky way in which laws are),
>> and to avoid possible hassle down the road from an unfriendly ILEC or an
>> upset AT&T who wanted to press the issue, it was decided that DIDs would
>> be purchased and assigned to those UNE circuits as they were deployed.
>> 
>
> I'm not sure I follow.  Voice trunks need routing to E911 tandems, but 
> what do data circuits have to do with this?
>   

Nothing. This is part of a law governing who can get UNEs. I don't have
it handy, but I'll look it up on Thursday (when I get back to the
office... have it in email there but not here for some reason).
>   
>> This is where we came in, and where the middle-man model still works to
>> some degree. They could simply buy great swaths of DIDs for themselves
>> at ridiculously low rates (being a LEC), but the caveat there is that
>> the DIDs have to be USED, or they're reassigned. 
>> 
>
> Depends on the area;  NANPA and pooling blocks aren't necessarily cheap.
>   

The numbers they quoted us were reasonable. Something like $500 for 2000
DIDs. Or possibly $200. Again... fuzzy on the exact numbers there, but I
remember it was quite good.
>   
>> We stepped in to
>> provide DIDs (which we purchase elsewhere) to their UNE circuits and
>> maintain them (even with no use), as well as maintaining the information
>> for E911 dispatch on each of the circuits (assuming, for the sake of
>> argument, that someone were to convert the data line into voice). Thus,
>> they can get the rates they want on the UNEs they deserve, and not worry
>> about the hassles of actually dealing with the technology and contracts
>> on the voice side that is simply not part of their core business model.
>> 
>
> Why would they have to deal with this when someone buying directly from 
> AT&T off the special access tariff doesn't?  (i.e. independent ISPs)
>   

Again. Thursday I'll have that info.
>   
>> Now this is, to be certain, an odd and unusual case. I doubt we could
>> find too many customers if that were our ONLY sort of business. But it
>> does illustrate your point that there is still, for now, a logical place
>> for the middle men companies in some situations.
>> 
>
> Agreed, although I'm still very confused as to why you need DIDs for 
> data UNEs.  Is this some bizarre feature of their ICA or something?
>   
It has to do with a recent modification of the telecom laws concerning
who's allowed to have access to the UNEs and who isn't, and it
stipulates that, in order to have access to them, you're now required to
be able to provide E911 service over them (as the law seems to just
outright assume that you'll be using them for voice). The law itself
doesn't seem to take into account that there's even a possibility that
someone

Re: [asterisk-users] CDR Rewrite -- Questions to the users

[Steve Murphy]

On Tue, 2009-01-13 at 21:09 +0100, Benny Amorsen wrote:

> I wrote a really long email, but it hinged on one thing I need
> clarified...
> 
> tir, 13 01 2009 kl. 09:05 -0700, skrev Steve Murphy:
> 
> >  CDR1:  A -> B  start: e1a  ans: e2  end: e4   Party: B  disp:
> > ANSW   linkedID: abc9
> >  CDR2:  A   start: e1   ans: e1  end: e6   Party: A  disp:
> > ANSW   linkedID: abc9
> 
> 
> We are talking about the "Simple" CDR's, not the leg-based ones,
> right? If so, why do all the CDR's only call one "Party"? Shouldn't
> there be a src and a destination? 
> 
> 
> /Benny
> 

Benny--

First, yes, all the examples I gave were for Simple CDR mode.

Next, the notation is simplified. A -> B  means channel: A   dstchannel:
B   (A and B are also contractions for stuff like "Dahdi/1-1" or
"Sip/bob-1")
start, ans, end are the 3 times (e2, e3, etc are event times (symbolic);
disp is the disposition field. linkedID is described in the doc.

I don't specify src/dest, in the examples, as they really don't convey
much without all the background description,
(if I said src = '101' and dest = '202' you'd have to know the
associated contexts, etc. -- a can of worms)---
but every CDR will specify those fields for the Dial (or whatever else
was responsible for activating the channel).

I also didn't spell out stuff like userfield, amaflags, callerid fields,
etc; it isn't that they aren't important, it just
clouds the examples to get too explicit and verbose.

The main thing about these fields is that they are in the list of CDR
fields and described in the CDR field section.

The "Party" field says which channel this CDR applies to. I use it
because the channel/dstchannel aren't always
going to involve the channel in question.  Usually, tho, Party will
either be the channel or the destchannel. Knowing
which one is the trick.

And, as a side note, the A CDR in my examples usually lacks a dstchannel
field, because it's not being activated by a dial--
it's being activated via an incoming call on an FXO interface. (or an
incoming SIP invite, maybe...)

And, if I'm missing info that would really, really, necessary to have,
or hard to inject from the dialplan, now is the time
to fight to make sure that it is in the spec.

murf

-- 
Steve Murphy 
Digium


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[asterisk-users] FWD and Asterisk

[David @ULC]

If I use below code in my sip.conf ,
[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very<<< this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no

how will call understand that where I have to land as we DO NOT provide our
IP in fwd configuration when we create an account.


>
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[asterisk-users] FWD and Asterisk

[David @ULC]

 When I logged in to my IPKall website ,

I see SIP Proxy: as fwd.pulver.com Do I need to change it to my PUBLIC or
STATIC IP ?
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Re: [asterisk-users] What are the various models of DID providers

[Alex Balashov]

SIP wrote:

>> I don't know that the price of UNE DS1s and DS3s is really all that 
>> exceptional.  Sure, it seems impressive that you can get a T1 in LATA 
>> 438 for some odd $44, but once you factor in the costs of 
>> interconnection, CO colocation, EELs and interoffice mileage if not 
>> colocated in the CO to which the circuit is being generated, private 
>> SONET for backhaul, etc.
 >
> The interconnection, CO colocation, private SONET, etc, are already in
> place in something like 60 municipalities and 4 Atlanta metro areas.
> They're using the UNEs to cut costs. Honestly, you could ask me some
> complex questions about their network, but I don't know it all that well...

I understand.

All I was trying to say is that the amortised cost of providing UNEs can 
be substantial, and that raw UNE prices on a per-circuit don't reveal 
that picture.

That doesn't mean it's not ultimately cheaper than going the 
wholesale/special access tariff route as a non-CLEC ISP would do.

>> I'm not sure I follow.  Voice trunks need routing to E911 tandems, but 
>> what do data circuits have to do with this?   
> 
> Nothing. This is part of a law governing who can get UNEs. I don't have
> it handy, but I'll look it up on Thursday (when I get back to the
> office... have it in email there but not here for some reason).

I'm not questioning your impression, but that doesn't make any sense, 
and I haven't heard that before.

What if you operate the kind of CLEC that just DACSs their CO CFA off 
their interconnect or private fiber into a data network full of circuit 
aggregation routers?

In other words, what if you don't even run a switch or provide voice of 
any kind?  What are you supposed to do then?  What if you're a "DLEC" 
like Covad?

>>   
>>> This is where we came in, and where the middle-man model still works to
>>> some degree. They could simply buy great swaths of DIDs for themselves
>>> at ridiculously low rates (being a LEC), but the caveat there is that
>>> the DIDs have to be USED, or they're reassigned. 
>>> 
>> Depends on the area;  NANPA and pooling blocks aren't necessarily cheap.
>>   
> 
> The numbers they quoted us were reasonable. Something like $500 for 2000
> DIDs. Or possibly $200. Again... fuzzy on the exact numbers there, but I
> remember it was quite good.

It really depends on the area.  But number resources are scarce;  to a 
large extent there is a similar problem there as with IPv4 address space 
in urban areas like Atlanta.

Pooling has helped a lot with that, though.  Now folks that can't 
justify a 10,000 block can still get numbers.

> It has to do with a recent modification of the telecom laws concerning
> who's allowed to have access to the UNEs and who isn't, and it
> stipulates that, in order to have access to them, you're now required to
> be able to provide E911 service over them (as the law seems to just
> outright assume that you'll be using them for voice). The law itself
> doesn't seem to take into account that there's even a possibility that
> someone might use a UNE for ONLY data (like many of the more recent
> modifications to the telecom act, it appears to have been hastily and
> vaguely written).

Which modifications?  (Yes, I know, you'll have the info when you're 
back at the office.  :-)

Are you sure this applies to T1s and such, rather than POTS / analog 
circuits that fall into the UNE spectrum?

I just went and looked at some ICAs of CLECs I've worked with and don't 
see anything about that for UNE circuits as such.  But, of course, these 
documents are incredibly recondite and hundreds of pages long.

-- Alex

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] What are the various models of DID providers

[Jai Rangi]

Alex,
I must say "wow", great explanation. It was a wonderful reading.
Best,
-Jai


On Tue, Jan 13, 2009 at 1:49 AM, Alex Balashov wrote:

> Hi Randulo,
>
> I think this topic is probably more appropriate for asterisk-biz, as was
> the aforementioned rant about one particular DID provider.  But,
> whatever - it is what it is.
>
> I assume that by "DID providers" you are referring to "origination" -
> that is, picking up calls on PSTN numbers and converting them to VoIP
> media and signaling and sending them to someone who wants to get numbers
> that ordinary PSTN users can call on a VoIP system of some kind.  The
> reason for the disambiguation is that many "DID providers" also provide
> "termination" - that is, the delivery of calls from VoIP into the PSTN.
>  There are also many companies that specialise in only origination or
> termination.  The two are closely related from a technical perspective
> but are characterised by rather different economics.
>
> At the end of the day--on a technical and a regulatory level--telephone
> numbers can only belong to a carrier.  A carrier is a network operator
> that is interconnected with other carriers and operates some form of
> switch, and usually interfaces via SS7 (or CSS7, as it is known outside
> North America) to the other carriers that they connect to.
>
> (Aside/digression about carriers:
>
> Of course, there are different types of carriers, depending on the
> jurisdiction.
>
> In the US, there are - broadly speaking - two different types:
> "incumbents" and "competitive" carriers involved in local service.
> Incumbents are either Bell system entities that were divested from the
> former AT&T monopoly in 1984 when AT&T was ordered to break itself up by
> the federal government, or various local-yokel independent telephone
> companies that were never acquired by AT&T during the 20th century (as
> well as various types of conglomerates that have bought some of these
> independents before, or since divestiture).  The latter type of
> incumbent is usually in small towns and/or rural areas, whereas the
> former is prevalent in metropolitan areas.
>
> The defining feature of an incumbent is that it tends to own the
> physical plant related to local telephone service delivery in a given
> area -- copper, fiber, central offices ("telephone exchanges"), remote
> terminals, junction boxes, conduit, and so on.   That's why it's an
> "incumbent."
>
> Examples of incumbents in the US include the former BellSouth (now
> AT&T), Ameritech, Qwest, Southwestern Bell (now AT&T), Verizon, GTE (now
> Verizon), and so on.  Independent incumbents include something like
> Ellijay Telephone Company here in Georgia, or Windstream (formerly
> Alltel).  This space has undergone a dizzying array of consolidation in
> the postmillenial years, so keeping accurate track of who is who even
> for pedagogical purposes is difficult.
>
> The Telecommunications Act of 1996 created "local loop" competition in
> the US and introduced the category of "competitive" carrier, or a CLEC
> (Competitive Local Exchange Carrier).  These are carriers that can
> interconnect with the incumbent (and in fact, the incumbent is legally
> required to interconnect with them) and have the right to lease certain
> parts of the incumbent's infrastructure at regulated rates in order to
> provide subscriber services - this pricing and resale discipline is
> known as UNE (Unbundled Network Element) in the parlance.  For example,
> a CLEC here in Atlanta in former BellSouth territory (now AT&T) connects
> their network to BellSouth and can rent the copper going back to my
> residence from BellSouth and generate all the services, features and
> routing from its own equipment and use BellSouth's plant to reach me
> over the "last mile."  CLECs can do other things as well;  they have
> various rights-of-way that let them build private networks across
> conduits in public spaces, they can lease dark fiber laid by electrical
> and gas utilities, etc.  But the defining feature of a CLEC is that they
> don't own the existing physical plant in place before, although they are
> welcome to overlay their own - in fact, that was very much the point of
> the Telecommunications Act.
>
> Most CLECs are small, but some are quite large and have a regional,
> national and even international footprint.  Examples of the large ones
> include Level3, Global Crossing, XO, McLeod USA, Paetec, Nuvox, etc. --
> these network operators all have CLEC status in many different
> incumbents' operating areas, if not necessarily all of them.
>
> Some CLECs neither do UNE nor really build networks nor lease anything,
> but exist for some specialised purpose to reap some economic or
> logistical advantage, like supporting the back side of a VoIP product or
> providing dedicated private transport between various large
> interconnection / peering points.  There are many different niches for
> the sort of thing that they are.  Nor does a CLEC have to have

Re: [asterisk-users] 404 not found from one ip-adress

[Ralf Träskman]

Thanks Your tip got my on the right track

Regards
/ralf

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristian 
Kielhofner
Sent: den 13 januari 2009 16:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 404 not found from one ip-adress

On Tue, Jan 13, 2009 at 9:59 AM, Ralf Träskman  wrote:
> Hi
>
>
>
> The provider dont use register, they are running openSER I have this in my
> sip.conf
>
>
>
> [outgoing]
>
> context=ip-only
>
> disallow=all
>
> allow=alaw,ulaw
>
> canreinvite=yes
>
> dtmfmode=rfc2833
>
> host=sip.hub.ip-only.se
>
> insecure=very
>
> reinvite=yes
>
> type=friend
>
>
>
> [incoming]
>
> disallow=all
>
> allow=alaw,ulaw
>
> context=ip-only
>
> type=user
>
>
>
> Regards
>
> /ralf
>

Ralf,

  That incoming peer isn't matching anything.

  They're probably hitting the context defined in [general].  Add
another peer/friend match with the other servers IP/hostname and the
ip-only context.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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