Re: [asterisk-users] The Good, Bad and Scam VoIP Providers

2006-12-27 Thread Kevin Walsh
Dovid B [EMAIL PROTECTED] wrote:
 A PI that does asterisk on the side ?? WTF ??
 
Do you have a list of business types that are not allowed to use VoIP?

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Re: [asterisk-users] Need quality toll free 800 number over IAX?

2006-12-23 Thread Kevin Walsh
Vicky [EMAIL PROTECTED] wrote:
 I have 3 toll free did's with nufone since 1 month .. Until now i dont have
 a problem with them .. their portal was good enough to do proper
 configuration and call quality wasnt bad  ( even though i havent used them
 in really huge traffic  yet ) .
 
The reason for the warning is to get people to move before they start to
have problems.  I'm sure everything will work for a while (perhaps for
years), but when you have a problem - perhaps an sudden deterioration or
complete outage at NuFone itself, you'll find out just how difficult it
is to get a helpful, or even just polite, response from their support
department.

You might find that you have no service through no fault of your own,
and have no way to fix it yourself.  Usually a threat of fix it or I'll
take my business elsewhere tends to work, but if you do follow up on
that then NuFone will rob you of any outstanding call credit you have
in your account.

You'll find lots of examples of this if you search around.  I ignored
the warnings because the service was working at the time, and I didn't
think I'd ever need to contact their support department.  Don't make
the same mistake.

Here are a few references I found with a five-minute Google search.
There are lots more where these came from:


https://forum.voxilla.com/other-providers/nufone-severe-problem-service-9548.html
http://www.voip-info.org/wiki/view/Nufone
http://www.shakataganai.com/index.php?/archives/163-NuFone.html
http://hansgrueber.blogspot.com/2006/01/nufone-sucks.html

There are loads of examples in the various mail list archives.
Here are a couple:

http://lists.digium.com/pipermail/asterisk-biz/2004-December/001468.html
http://lists.digium.com/pipermail/asterisk-biz/2004-December/001457.html

As you have only had the DDIs for a month, you could get away with simply
going elsewhere and using new numbers (use up your call credit first).
If you have vanity numbers, or you want to keep them for some other reason
then I wish you good luck getting NuFone to help you port them to your
new provider.

The bottom line is that you are free to use whichever provider(s) you
like, but please be aware of the high likelyhood of having severe
service difficulties with NuFone.

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Re: [asterisk-users] Need quality toll free 800 number over IAX?

2006-12-22 Thread Kevin Walsh
Andrew Joakimsen [EMAIL PROTECTED] wrote:
 NuFone isnt bad if you want a disposable termination account. But don't rely
 on it for anything.
 
Well, the voice quality left a lot to be desired, so I didn't make a
lot of use of the service anyway.  Perhaps, if I had made more use of
it, they wouldn't have been able to steal as much from me when I fired
them for incompetence, laziness and general rudeness.

Perhaps they think that if they rob enough people then they will be
in a better position to pay their upstream provider - leading to less
downtime and therefore less cause for customers to attempt to contact
the idiot running their support department (Jeremy).

They are in urgent need of a better business plan (one that doesn't
rely upon raising money by stealing funds from customer accounts) and
a complete change of staff.  Oh, and a clue!

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Re: [asterisk-users] Need quality toll free 800 number over IAX?

2006-12-22 Thread Kevin Walsh
Doug Crompton [EMAIL PROTECTED] wrote:
 If this (or any) company is really stealing or not living up to a contract
 then why not report them as such, especially if they are US based. I would
 suspect you would have another route to take.
 
I am UK-based - there's not a lot I can do to a US-based company,
and they know it.  That is unless I'm willing to fly over there to
sue them, which I'm not.  In this case, it was better to just cut
our losses and make sure as many others as possible know about our
experience and are forewarned.

I have already convinced several others to abandon NuFone.  I was
the one who recommended them in the first place, so I felt that I
had to warn them and try my best to persuade them to migrate.

Remember - sometimes, when you annoy one customer, you loose a lot
more than just that one customer.  Whatever playground victory NuFone
thinks they won by helping themselves to the content of our account,
they lost hundreds of times over in terms of the future revenue from
the customers I know that they lost.


 If you don't do anything about it then they will just go on abusing
 others and getting away with it.

That's why I feel it is right and proper to warn others.

If NuFone carries on operating they way they do, they will lose a
lot more than just the customers we had powers of persuasion over.
The snowball effect induced by mounting bad publicity is a powerful
thing, and not something any company wants to be on the receiving end
of.

I know that I have caused more damage this way than I could ever do by
simply recovering the account balance in court, and I didn't have to fly
anywhere to do it.  NuFone's short-sighted and clearly criminal ways
will come back to haunt them one day.

 
 At the very least the BBB (bbb.org) should be notified. They have a web
 site and if it is really wire/internet fraud then the FBI
 (www.fbi.gov/majcases/fraud/internetschemes.htm) has a site you can
 register a complaint with.

I've never heard of the BBB.  I have now - thanks.  I doubt that
NuFone's behaviour counts as fraud though.  I'd class their actions
as just plain old-fashioned theft.

If you are a NuFone customer then I advise you to use up your balance
and leave as soon as possible.  It's very easy to do - especially if
you're only using a company as an outgoing route and don't need to port
a number to a new provider.  If you know any NuFone customers then you
should try to get them to do the same.

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Re: [asterisk-users] Need quality toll free 800 number over IAX?

2006-12-22 Thread Kevin Walsh
Doug Crompton [EMAIL PROTECTED] lazily top-posted:
 Well if this company is US based I would not think where you are matters
 if it is fraud. You could still enter a complaint at the FBI site.
 
 I would also think they would be working with the UK counterpart.
 
I doubt it - not on a minor theft case, perpertrated by minor criminals.

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Re: [asterisk-users] Need quality toll free 800 number over IAX?

2006-12-20 Thread Kevin Walsh
www.IPKall.com [EMAIL PROTECTED] wrote:
 I need a quality US 800 DID over IAX for my Asterisk server, preferably one
 that doesn't cost the earth.
  
 Any suggestions please?
  
Anyone except NuFone.

Their customer service is non-existant - you have to email every day
for a couple of months before you'll be privileged enough to get a
one-line response to a service outage issue.  If you dare to point
out that the response didn't address the issue then you'll unleash the
combined wrath of both of the brain cells in residence at NuFone's
support department.  Not immediately, of course - you'll have to wait
another couple of months for a reply.

If you give up on them and decide to go elsewhere, they will pocket any
outstanding funds you have pre-paid into your account.  Existing
NuFone customers are advised to not pre-pay too much to these yokels,
and to jump ship as soon as possible.

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RE: [Asterisk-Users] Largest working config files?

2005-10-24 Thread Kevin Walsh
Steve Davies [EMAIL PROTECTED] wrote:
 I hope this is not a FAQ - I have not been able to find it if it is
 covered already... 
 
 I have a dial-plan on my asterisk system that is becoming potentially
 quite large and complex - Of the order of 12 lines of dialplan per
 extension number. Most of this is in order to record suitable CDR
 data, access voicemail, and play polite goodbye messages etc. The
 operation of each extension can potentially be unique, making a common
 [extensions-generic] almost impossible to write.

Have you looked into creating a couple of macros to reuse your code?

[local-extensions]
exten = 2100,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN})
exten = 2101,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN})
exten = 2102,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN})
exten = 2103,1,Macro(call-local,${EXTEN},cursor,${OPERATOR_EXTEN})
...

As you can see, I only need one line per extension;  All of the call
logic is in the [macro-call-local] macro.  The maintenance is a lot
simpler too, of course.

 
 Does anybody have experience of how big an extensions.conf can get
 before problems start occuring? If anyone has experienced problems, what
 sort of things happen? 
 
I have no idea.  Here's our dialplan line count (quite small because of
the macros):

 218 extensions/incoming.conf
 354 extensions/internal.conf
 225 extensions/macros.conf
 471 extensions/outgoing.conf
 151 extensions/routes.conf
1419 total

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RE: [Asterisk-Users] Re: www.openpbx.org

2005-10-07 Thread Kevin Walsh
Brian C. Fertig [EMAIL PROTECTED] wrote:
  Further info.  The domain is registered to Marc Olivier Chouinard.  He
  has posted in the dev list. 
  
 Can they do this?   Is this legal?
 
Yes - anyone can register a domain name.

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RE: [Asterisk-Users] Re: www.openpbx.org

2005-10-07 Thread Kevin Walsh
Jon Pounder [EMAIL PROTECTED] wrote:
 There are people out there who wish to contribute, and not have their work
 lost on an individual project website since they do not choose to accept
 digium's terms to contribute to asterisk. This gives them an opportunity
 to do so, and have their work aggregated with everyone else in the same
 category, so it is one stop shopping for users.
 
 Open source is about choices, not restrictions, and this gives
 contributors more choice. As long as the two streams stay compatible
 (which they likely will) it should be better for everyone.
 
I think it's a great idea, and long overdue.

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RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents

2005-10-06 Thread Kevin Walsh
Matt Riddell [EMAIL PROTECTED] wrote:
 My comment was directed at the USPTO who grants patents on a
 regular basis with what seems like no effort to check for prior art.  The
 first time I saw this I thought it was stupid.  The second, unbelievable,
 and the 358456347563th one crazy! 
 
The life of a US patent clerk must be so boring.  The greatest excitement
they probably get is when they run out of ink for their approved
rubber stamp.  The USPO could probably outsource their approval process
to a third-world sweat shop.

If patent clerks were forced concentrate on their work, there would be
no atomic bomb.

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RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents

2005-10-05 Thread Kevin Walsh
trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote:
 Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing
 on VoIP patents.  Sprint Nextel claims to have about 100 patents on VoIP
 technologies.  Does anyone know which ones this article is talking
 about, and if so does asterisk have any of those features?
 
 The reason I am asking is that the article is vague, Vonage uses a
 fairly standard codec set, I dont know about the others.  So if its not
 codecs I wonder if its something so generic that the patent would be
 tossed out upon challenge. 
 
 Anyone thinking about doing a VoIP business may want to get more info
 before proceeding since they may not have the millinos vonage has to
 fight this. 
 
Marvellous.  Another company with a monopoly over aspects of VoIP
technology.  I don't have the millions required to mount a defence
in a North American court, so I should just consider myself lucky that
I live in a free country.

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RE: [Asterisk-Users] Intel Pentium Celeron

2005-10-05 Thread Kevin Walsh
Giordano Grandis [EMAIL PROTECTED] wrote:
 (Article auto-converted from unnecessary HTML to nice plain text.)

 i'm going to install asterisk with a 4 BRI (HFC chipset) on a Celeron at
 2.6 GHz I don’t known Celeron performance, but i listen that is not very
 good.

 Could I have some performance isuue with this kind of processor ?

You could have performance issues with any processor;  It all depends
upon what you want to do with it.

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RE: [Asterisk-Users] inter Asterisk trunking IAX /IAX2

2005-10-05 Thread Kevin Walsh
Geo [EMAIL PROTECTED] wrote:
 Anyone using inter Asterisk trunking IAX /IAX2 ?

No - you're the first to think of that.  Congratulations.

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RE: [Asterisk-Users] IPComms Setup

2005-10-05 Thread Kevin Walsh
Crystal Stream, Incorporated [EMAIL PROTECTED] wrote:
 Hey

Moo.


 I just setup service with IPComms and they are
 telling me to setup such as this:
 
 iax.conf:
 [IPCommsNet]
 type=user
 host=69.15.xxx.xx
 context=voicepulse-in ;(changed by me)
 nat=yes
 dtmfmode=rfc2833
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 
 When I'm calling once of my numbers it's giving me
 this though:
 
 Oct  5 12:11:06 NOTICE[49584]: chan_iax2.c:5476
 socket_read: Rejected connect attempt from
 69.15.xxx.xx, request '[EMAIL PROTECTED]' does
 not exist
 
Do you have a context called [voicepulse-in] containing a definition
of what to do when someone calls that DDI number?

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RE: [Asterisk-Users] IPComms Setup

2005-10-05 Thread Kevin Walsh
Crystal Stream, Incorporated [EMAIL PROTECTED] wrote:
 this isn't working
 [IPComms-in]
 exten = s,1,Noop(${DATETIME} ${CALLERID})
 exten = s,2,SetCallerID(${CALLERID})
 exten = s,3,Answer
 exten = s,4,Goto(main-menu,s,2)
 exten = s,5,Hangup
 
 What I have is a block of 20 DIDs and I want to accept calls from all of
 them. 
 
 It would be way to freaking complicated to do
 exten = 2027575120,1,Noop( 
 .
 exten = 2027575121,1,Noop( 
 et cetera
 
 How do I get this done?
 
You could wildcard your DDIs, replacing 20, 21 etc. with [23][0-9], or
whatever.  Alternatively, you could create a macro that would look
a lot like the body of your [IPComms-in] context, and then call that
from 20 separate DDI exten lines.  I'd just go with the wildcard.

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RE: [Asterisk-Users] Answering Machine Detection

2005-10-05 Thread Kevin Walsh
 Anyone aware if Digium or Sangoma, or possibly a function of Asterisk,
 supports answering machine detection on an outbound call?

I usually just listen to the voice.  If it says something along the
lines of leave a message after the tone then it's probably a machine.
I can then choose to leave a message or just hang up.

I imagine that most people would take a similar approach, even though
the steps are probably not documented anywhere.

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RE: [Asterisk-Users] Voicemailmain automatic extension detection?

2005-10-05 Thread Kevin Walsh
Mason Loring Bliss [EMAIL PROTECTED] wrote:
 Is there a way I can have voice mail check calls coming from my internal
 users automatically get to the right extension, without having the user
 enter their extension? 
 
 I'm thinking that I could have the local SPA boxes translate, or have
 each user live in a context where the extension in question exists
 uniquely per user, but both of these seem kludgey.
 
 Thanks in advance for clues!

Do you mean something like VoiceMailMain(${CALLERIDNUM})?

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RE: [Asterisk-Users] G.729 patent in France

2005-09-30 Thread Kevin Walsh
Amaury BOSSE [EMAIL PROTECTED] wrote:
 I am building an Asterisk PBX with voicemail and music on hold functions.
 An ISDN BRI line will also be available and G.729 IP-phones will be used.
 
 Are there patents rights applicable to France?
 
The European Parliament recently voted 648 to 14 to reject the Computer
Implemented Inventions Directive.  The directive was supported by large
monopolists such as Microsoft and would have thrown us into the same
software patent minefield as the USA.

The defeat of the bill means that individual EU member countries will
continue to make their own decisions on what is patentable, rather than
being hamstrung by the proposed EU-wide bill.  Software-only patents are
not valid in England and probably not in France, although that's for you
to check.  I understand that a limited number of software patents are
valid in Italy.

Software is protected by copyright, and that's enough.  Ideas are free.

http://www.nosoftwarepatents.com/

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RE: [Asterisk-Users] G.729 patent in France

2005-09-30 Thread Kevin Walsh
Steve Underwood [EMAIL PROTECTED] wrote:
 A large percentage of the patents applicable to G.729 are held by France
 Telecom. Now guess whether they bothered to get those patents in France.

British Telecom has a large number of patents in North America.  It can't
use its software-only patents in England, of course.

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RE: [Asterisk-Users] FW: defunct email kill list

2005-09-02 Thread Kevin Walsh
Dean Collins [EMAIL PROTECTED] wrote:
 (Article auto-converted from unnecessary HTML to nice plain text.)

 Is there an email address that we can forward 'defunct' emails like the
 one below to so that they can be taken off the mailing list. I appreciate
 it means work for someone but this email is probably being sent out to
 400 'original' posters a day if that makes sense.

 It could also be the email address that we forward out of office
 emails to that could automatically suspend them for a period of time.

 You could probably automate the service but wouldn’t be hard to have
 a person read the email, copy the address and paste it into a webpage.
 Just a thought.

It seems to me that an automated solution to this problem would leave
itself open to abuse.  A moderated system would have to be monitored
by someone, and that would seem to be far too much of a chore.

You'll probably find that people with the appropriate access will
unsubscribe or suspend accounts that send them OoO messages anyway,
without further prompting from other subscribers.

Perhaps your proposed auto-unsubscribe feature should be reserved for
people who insist upon sending HTML emails and/or lazily top-posting
their followups. :-)

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RE: [Asterisk-Users] [OT] Looking for Web based SIP endpoint

2005-08-20 Thread Kevin Walsh
Matt Riddell [EMAIL PROTECTED] wrote:
 Jason Becker wrote:
  https://sip-communicator.dev.java.net/
  
  Don't know the current state of functionality with Asterisk. I couldn't
  get it to work many months ago - even with help from the developer.
 
 Any reason you are looking for SIP and not IAX?

Is there an IAX alternative that you'd recommend?

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RE: [Asterisk-Users] Installing to a prefix.

2005-08-20 Thread Kevin Walsh
Trevor G. Hammonds [EMAIL PROTECTED] wrote:
 Kevin Walsh wrote on Friday, 19 August 2005 6:58 PM:
  Andrew Kohlsmith [EMAIL PROTECTED] wrote:
   May I humbly request they be attached to a feature request on Mantis?
   
  I've been less than humbly requested not to do that sort of thing any
  longer, as I haven't signed a disclaimer.  Sorry about that.
  
  The Asterisk change is trivial;  Just set the INSTALL_PREFIX variable
  in the Makefile and then modify asterisk.conf and possibly
  musiconhold.conf. The Zaptel Makefile changes are a bit more
  involved.  The diff file is 148 lines long.  I've never had cause to
  look at libpri.
 
 How about submitting a disclaimer to Digium for the modified makefiles?
 
Let's not bring that subject up again. :-)

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RE: [Asterisk-Users] Installing to a prefix.

2005-08-20 Thread Kevin Walsh
Tzafrir Cohen [EMAIL PROTECTED] wrote:
 On Sat, Aug 20, 2005 at 02:57:50AM +0100, Kevin Walsh wrote:
  Andrew Kohlsmith [EMAIL PROTECTED] wrote:
   On Friday 19 August 2005 21:27, Kevin Walsh wrote:
I'll send the modified Makefiles to anyone who needs them.

   May I humbly request they be attached to a feature request on Mantis?
   
  I've been less than humbly requested not to do that sort of thing any
  longer, as I haven't signed a disclaimer.  Sorry about that.
  
  The Asterisk change is trivial;  Just set the INSTALL_PREFIX variable
  in the Makefile and then modify asterisk.conf and possibly
  musiconhold.conf. The Zaptel Makefile changes are a bit more involved. 
  The diff file is 148 lines long.  I've never had cause to look at
  libpri. 
 
 INSTALL_PREFIX/DESTDIR is not instended for that. It is intended for
 installing asterisk to a different prefix than the one you build it to.
 It is commonly used for building installation packages (e.g: for rpms or
 debs). 
 
That's fine for Asterisk.  The Zaptel Makefile, as distributed, doesn't
play nicely if you change the prefix.

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RE: [Asterisk-Users] Installing to a prefix.

2005-08-19 Thread Kevin Walsh
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
 On Friday 19 August 2005 21:27, Kevin Walsh wrote:
  I'll send the modified Makefiles to anyone who needs them.
 
 May I humbly request they be attached to a feature request on Mantis?
 
I've been less than humbly requested not to do that sort of thing any
longer, as I haven't signed a disclaimer.  Sorry about that.

The Asterisk change is trivial;  Just set the INSTALL_PREFIX variable
in the Makefile and then modify asterisk.conf and possibly musiconhold.conf.
The Zaptel Makefile changes are a bit more involved.  The diff file is
148 lines long.  I've never had cause to look at libpri.

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RE: [Asterisk-Users] re: call load balancing

2005-08-12 Thread Kevin Walsh
Anton Krall [EMAIL PROTECTED] wrote:
 [EMAIL PROTECTED] intruder]# ps afx|more
   PID TTY  STAT   TIME COMMAND
 1 ?S  0:08 init
 2 ?SW 0:00 [keventd]
 3 ?SW 0:00 [kapmd]
 4 ?SWN0:00 [ksoftirqd_CPU0]
 9 ?SW 0:00 [bdflush]
 
 No priorities.. Am I missing something?
 
Try ps alx (Look at the NI column).

Also see man ps.

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RE: [Asterisk-Users] *** Asterisk 2.0 Stable release out now

2005-08-11 Thread Kevin Walsh
Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
 Justin Selleck wrote:
  Is asterisk 2.0 real?  Running in c#?  I see references to it but cannot
  find it anywhere.
 
 r: Generate a ringing tone for the calling party, passing no audio from
 the called channel(s) until one answers. Use with care and don't insert
 this by default into all your dial statements as you are killing call
 progress information for the user. Really, you almost certainly do not
 want to use this. Asterisk will generate ring tones automatically where
 it is appropriate to do so. r makes it go the next step and
 additionally generate ring tones where it is probably not appropriate to
 do so.

I think you might have replied to the wrong article.

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RE: [Asterisk-Users] re: call load balancing

2005-08-11 Thread Kevin Walsh
Joseph [EMAIL PROTECTED] wrote:
 On Wed, 2005-08-10 at 09:11 -0700, 1 2 wrote:
  
  I run asterisk with the -p option instead of messing with nice levels
  and it seems to make an improvement.
  
 If asterisk starts from the script how to append -p option.
 
 This is command from the script that starts asterisk:
 start-stop-daemon --start --exec /usr/sbin/asterisk \
 ${OPTS} -- ${ASTERISK_OPTS}
 
 adding -p doesn't work:
 start-stop-daemon --start --exec /usr/sbin/asterisk -p \
 ${OPTS} -- ${ASTERISK_OPTS}

Put the -p in ${ASTERISK_OPTS} or at the end of that line.

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RE: [Asterisk-Users] call load balancing

2005-08-10 Thread Kevin Walsh
Joseph [EMAIL PROTECTED] wrote:
 On Wed, 2005-08-10 at 08:10 +0400, Jean-Michel Hiver wrote:
  I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable
  difference in call statistics (i.e. avg length of calls). If you are
  using ADSL, the maximum bandwith you'll be able to use is your upload
  rate since VoIP calls send data bidirectionally.
  
  Of course if you're using g.711 it's a different kettle of fish since it
  takes 80kbps (g.729 only uses about 24).
  
 According to Wiki:
 G729 is 8Kbps
 G711 is 64Kbps
 http://www.voip-info.org/tiki-index.php?page=Codecs

That's the payload.  You need to add the IP overhead to those numbers,
which will bring the total to what Jean said, above.  Trunking your
calls over an IAX link will help reduce the total IP overhead.

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RE: [Asterisk-Users] re: call load balancing

2005-08-10 Thread Kevin Walsh
Joseph [EMAIL PROTECTED] wrote:
 Don't forget to experiment with nice to increase priority of for
 Asterisk.
 By default asterisk run with priority 0 same as apache and any other
 applications.
 We run a web-server on the same machine as asterisk and increasing
 nice for Asterisk to -15 helped a lot.

You don't need to mess about with nice.  Just run Asterisk with realtime
priority;  Use the -p switch when you start the Asterisk daemon.

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RE: [Asterisk-Users] Full T38 sip Faxing now Available

2005-08-03 Thread Kevin Walsh
Carlos [EMAIL PROTECTED] lazily top-posted:
 Has anyone got a response from this?
 
It was just spam.  Forget it.

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RE: [Asterisk-Users] asterisk.org beta site up!

2005-08-03 Thread Kevin Walsh
Kristof Hardy [EMAIL PROTECTED] wrote:
 Matt Brooks wrote:
  I am just emailing to inform you guys that a new website has been
  created for asterisk.org.  You can find the beta site up at
  http://beta.asterisk.org.  It utilizes the drupal portal framework and
 
 Looking very good and much easier to navigate! Great work!
 
Well, at least the new website doesn't say that I can register today
to participate in an event that took place last June (see www.asterisk.org).

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RE: [Asterisk-Users] Full T38 sip Faxing now Available

2005-08-03 Thread Kevin Walsh
Chris Mason (Lists) [EMAIL PROTECTED] wrote:
 Kevin Walsh wrote:
  Carlos [EMAIL PROTECTED] lazily top-posted:
   Has anyone got a response from this?
   
  It was just spam.  Forget it.
  
 I have an account with them, just waiting for a suitable ATA to arrive.

Good for you.  Personally, I never buy anything from spammers.

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RE: [Asterisk-Users] Full T38 sip Faxing now Available

2005-08-03 Thread Kevin Walsh
Michael D Schelin [EMAIL PROTECTED] wrote:
 Why do you put me down? I have not done a thing to you and I'm not a
 spammer. Please stop this activity It's not professional. If I were to
 give you bad service please feel free to comment negatively but I've
 never dealt with you nor do you have an account with us.
 
As I understand it, you sent non-Asterisk-related commercial
announcement to the Asterisk Users' mail list.  What made you think
that that wouldn't be considered to be Spam?

I obviously can't comment on your service, as I'm unlikely to become
a customer.

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RE: [Asterisk-Users] [Asterisk-Dev] Digium to Sponsor a Pizza party atCluecon

2005-08-01 Thread Kevin Walsh
Brian West [EMAIL PROTECTED] wrote:
 Digium, the creator and primary developer of Asterisk, the industrys
 first Open Source PBX, will be hosting a pizza party from 4pm to 6pm
 on the first day of Cluecon. We look forward to everyone coming out
 to enjoy this opportunity to meet fellow developers and users in a more
 casual environment. 
 
Thanks.  I missed the first 600 copies of that announcement. :-)

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RE: [Asterisk-Users] help on linux version

2005-07-28 Thread Kevin Walsh
amna saleem [EMAIL PROTECTED] wrote:
 (Article auto-converted from unnecessary HTML to nice plain text.)

 This is amna saleem.I needed to ask if asterisk-1.0.3 can run on linux
 enterprise edition(latest version 4)

I assume you mean Red Hat Enterprise.  It should run on any modern
GNU/Linux distribution.  All of the servers under my control are
running Gentoo.

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RE: [Asterisk-Users] Public phone

2005-07-28 Thread Kevin Walsh
Chris Mason (Lists) [EMAIL PROTECTED] wrote:
 A client wants to put phones in a semi-public place, using a Calling
 Card solution. What kind of hardware is suitable? I'm looking for a wall
 mounted booth, a SIP phone that can't easily be broken, but I might just
 use cheap analog phones and a channel bank.
 What do you suggest for the calling card software?
 This installation will be outside the US.

Buy a payphone with an internal rate table and hook it up to a Sipura
SPA-2000, or a similar ATA box.  That'll save a lot of messing about
with calling cards etc.

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RE: [Asterisk-Users] Full T38 sip Faxing now Available

2005-07-28 Thread Kevin Walsh
Tom [EMAIL PROTECTED] top-posted:
 Why the big secret?  Why not post your solution to the list?
 
It's probably just another one of those nasty closed source add-ons
for sale.

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RE: [Asterisk-Users] Re: [Asterisk-Dev] Cluecon - Who's going ?

2005-07-26 Thread Kevin Walsh
Greg Boehnlein [EMAIL PROTECTED] wrote:
 I'll be REALLY interested in your talk! Please make sure that you have
 take-away notes available so it doesn't evaporate into thin air after the
 conference! :)

Perhaps also a drive-through lane at the side of the venue for people
who don't have the time to find a seat, let alone sit and take notes.
:-)

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RE: [Asterisk-Users] RE: Business Edition

2005-07-25 Thread Kevin Walsh
Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
 Kevin Walsh wrote:
  Brian West [EMAIL PROTECTED] wrote:
   Or better yet.. modify the disclaimer like I and a few others did to
   say that the only thing you will disclaim are things you post on the
   bug tracker!  NO UPDATES, NO CHANGES, NO NOTHING!  If its not posted
   under your user on mantis IT IS NOT DISCLAIMED!
   
  That would seem to be a reasonable suggestion.
  
 That's what I did.  I modified the disclaimer to only apply to stuff I
 post to bugs.digium.com under a specific userid.
 
 I did this to keep stuff I post to the mailing lists or on the web from
 being accidently disclaimed.

Most people probably are not aware that that's an option.  I certainly
wasn't aware of it.  If the owner accepts custom agreements, rather
than just one of the two published versions, then that's a good start.

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RE: [Asterisk-Users] RE: Business Edition

2005-07-24 Thread Kevin Walsh
Brian West [EMAIL PROTECTED] wrote:
 Or better yet.. modify the disclaimer like I and a few others did to
 say that the only thing you will disclaim are things you post on the
 bug tracker!  NO UPDATES, NO CHANGES, NO NOTHING!  If its not posted
 under your user on mantis IT IS NOT DISCLAIMED!
 
That would seem to be a reasonable suggestion.

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RE: [Asterisk-Users] RE: Business Edition

2005-07-23 Thread Kevin Walsh
Tzafrir Cohen [EMAIL PROTECTED] wrote:
 Disclaimers aside, who has the copyrights in those cases?
 
 Digium currently holds copyrights and/or is allowed to relicense the
 full asterisk codebase as is currently distributed in the asterisk
 tarballs on ftp.asterisk.org and also all the code in the asterisk CVS
 on cvs.digium.org (any better definition?) .
 
Just to be clear, the perpetual agreement doesn't force a transfer
of copyright;  The author gets to keep the copyright, and can do
whatever he likes with the code.  The shorter disclaimer puts the
copyright into the public domain.

The perpetual agreement gives the owner two main rights.  Firstly
paragraph 1 allows the code, and all future Asterisk-related code,
written by that contributor to be closed by the owner.  Secondly,
paragraphs 2 and 5(a) force the contributor to report all future
changes and/or enhancements to save the owner the hassle of having
to scour future forks looking for code that they might be interested
in folding into their proprietary release.

The second disclaimer (the short one) simply dumps all of your changes
and enhancements into the public domain for anyone to use in a
proprietary product.  Of course, the only people who would know about
this would be the signer and the company to which the document was sent.

The short disclaimer is sufficiently woolly to allow for all future
changes to a fork to be folded back into the binary release, although
it doesn't include an obligation to report all such changes.

Once signed, neither agreement has an exit clause or time limit, so
neither of them can be cancelled.  Maybe that is legal in Alabama
(or Delaware), but I wouldn't really want to have to travel there to
find out.

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RE: [Asterisk-Users] RE: Business Edition

2005-07-23 Thread Kevin Walsh
 On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote:
  Adam Goryachev [EMAIL PROTECTED] wrote:
   On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote:
For this reason, I believe that if a fork were
ever necessary, it would struggle to beat a distinct path away from
the Asterisk Binary Edition
   
   Correct, until the point where there is MORE features being added to
   the forked version of asterisk than the digium version of asterisk.
  
  That can't happen, because the ABE could, and probably would, absorb
  all of the advances in the fork, while forging ahead with the
  original.
 
 Since the fork would be GPL only, if ABE 'absorbed' the new features,
 then it would 'become' GPL, and therefore would need to be released as
 GPL, and hence would no longer by ABE :) So, that can't happen. Any other
 ideas? 

You're forgetting about the disclaimer documents.  Anyone who signed
the perpetual agreement and made changes and/or enhancements to the
Asterisk code (a fork would still be using Asterisk code) would firstly
be obliged to inform the owner, and would secondly have a prior
agreement with the owner to allow them to use and close the code.
That would neatly bypass the GPL and allow the new code to be folded
into the Asterisk Binary Edition.

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RE: [Asterisk-Users] RE: Business Edition

2005-07-22 Thread Kevin Walsh
Kevin P. Fleming [EMAIL PROTECTED] wrote:
 Kevin Walsh wrote:
  The perpetual agreement grants the owner a non-cancellable right
  to use changes and/or enhancements made to the Asterisk codebase as
  [the] owner sees fit.  As any Asterisk fork would, of course, be based
  upon existing Asterisk code, the owner would have the automatic right
  to take any code they wanted and backport it into the Asterisk Binary
  Edition - as long as the contributor to the fork had previously signed
  a perpetual disclaimer at some point in the past.
 
 Nice work clipping out only the words you wanted to use there! Let's try
 this again, with the actual text from the disclaimer:
 
 (b) The rights made in Para. 1(a) of this Agreement applies to all past
 and future contributions of Contributer that constitute changes and
 enhancements to the Program. 
 
 2.  Contributer shall report to Owner all changes and/or enhancements to
 the Program which are covered by this Agreement, and (to the extent known
 to Contributer) any outstanding rights, or claims of rights, of any
 person, that might be adverse to the rights of Contributer or Owner.
 
 In other words, the _only_ code that the disclaimer covers is that which
 the Contributer directly identifies to Digium to be covered by the
 disclaimer. In absolutely no way does this disclaimer give Digium the
 right to appropriate other changes the Contributer makes to the covered
 programs without their knowledge and permission.

Firstly, there are no in other words about it.  That is a legal
document.  If other words are meant then they should be stated as
such - in plain English.  Secondly, paragraph 2 is distinct from
paragraph 1, in which paragraph 2 insists that the contributor to a
fork to also report changes back to the owner.  If the owner doesn't
report changes then they could find themselves in trouble over a
breach of the agreement, and the owner can still simply take the
changes anyway, as allowed for in paragraphs 1(a) and 1(b).

 
 In addition, even the most liberal interpretation of these clauses still
 includes the words Contributer and contribution, which clearly means
 that the entity signing the disclaimer has sole discretion which of
 their changes are covered and which are not.

If that's the intention then it should be made clear in the document.
The agreement, as it stands today, contradicts your statement and I
believe it has been very carefully worded to either hide its true
intentions or to allow future loopholes in favour of the owner.

By the way, if anyone wants to see the full text of the dangerously
perpetual disclaimer, they can find it here:

http://www.digium.com/disclaimer.txt

Read it very carefully, or have a lawyer advise you as to its content
and implications.

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RE: [Asterisk-Users] RE: Business Edition

2005-07-22 Thread Kevin Walsh
Adam Goryachev [EMAIL PROTECTED] wrote:
 On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote:
  It has been flippantly said, a number of times, that if you don't
  like the situation then you can fork the project.  A major fork seems
  (to me) to be pointless for one main reason (and a couple of lesser
  reasons): 
  
  As I see it, anyone working on an Asterisk fork who had previously
  signed the dangerous disclaimer (the perpetual one) could find their
  changes to the fork rolled back into the Asterisk Binary Edition
  without any further permission being required.
  
  The perpetual agreement grants the owner a non-cancellable right
  to use changes and/or enhancements made to the Asterisk codebase as
  [the] owner sees fit.  As any Asterisk fork would, of course, be based
 
 IANAL, but I assume you also have the right to revoke the agreement as
 relating to future patches. ie, it is non-cancellable in that I can't
 contribute something today, and next week change my mind. I am sure I
 can sign the agreement, contribute enhancements, cancel my agreement,
 and no longer contribute enhancements.
 
That is not the case.  The agreement makes it clear that 1(a) the
signer does hereby grant, a non-exclusive, royalty-free and
non-cancellable right to use changes and/or enhancements made to the
programs. and 1(b) this Agreement applies to all past and future
contributions of Contributer (sic).

There is no provision to cancel, and furthermore, the signer
specifically agrees to this arrangement by signing.

 
  For this reason, I believe that if a fork were
  ever necessary, it would struggle to beat a distinct path away from
  the Asterisk Binary Edition
 
 Correct, until the point where there is MORE features being added to the
 forked version of asterisk than the digium version of asterisk.

That can't happen, because the ABE could, and probably would, absorb
all of the advances in the fork, while forging ahead with the
original.

 The *average* feeling of the community is that they are happy with
 the status quo.

The status quo has been disrupted with the unveiling of the Asterisk
Binary Edition.

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RE: [Asterisk-Users] RE: Business Edition

2005-07-22 Thread Kevin Walsh
Brian Capouch [EMAIL PROTECTED] wrote:
 Kevin Walsh and Aidan are able to see things that the rest of us cannot.
   Digium has duped you into associating with their evil enterprise to
 appropriate everyone else's hard work.
 
 I'm sure the stuff you and Mark have contributed pales in comparison with
 *their* contributions!! 
 
You'll never know.  Contributors are required to sign a disclaimer,
which is something I cannot do.

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RE: [Asterisk-Users] Re: RE: Business Edition

2005-07-22 Thread Kevin Walsh
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
 On Friday 22 July 2005 12:04, Lee Howard wrote:
  Well, I'm sure that was an added bonus.  :-)  Free work and free money.
  It reminds me of a certain Dire Straits lyric.
 
 Yes but are the chicks free?
 
All except for the binary ones.

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RE: [Asterisk-Users] RE: Business Edition

2005-07-22 Thread Kevin Walsh
Steve Underwood [EMAIL PROTECTED] wrote:
 Kevin P. Fleming wrote:
  2.  Contributer shall report to Owner all changes and/or enhancements to
  the Program which are covered by this Agreement, and (to the extent
  known to Contributer) any outstanding rights, or claims of rights, of
  any person, that might be adverse to the rights of Contributer or Owner.
  
  In other words, the _only_ code that the disclaimer covers is that
  which the Contributer directly identifies to Digium to be covered by
  the disclaimer. In absolutely no way does this disclaimer give Digium
  the right to appropriate other changes the Contributer makes to the
  covered programs without their knowledge and permission.
 
 Well, yes, that's the idea. However, those of us who contribute don't
 generally provide any clear traceable definition of what we are
 contributing and what we are not. The documentation here is pretty woolly.
 
As I pointed out, paragraph 2 doesn't have any bearing upon what code
the owner is allowed to incorporate into the Asterisk Binary Edition;
That non-cancellable right is granted in paragraph 1.

The quoted paragraph (2) simply forces the person who signed the
document to report any changes to the Asterisk codebase.  In theory,
even minor changes that would be of no use to the wider community must
be reported to the owner, who would make the final decision.

A fork followed by changes and/or enhancements to the code would be
covered by that paragraph - forever.  It's all very underhanded.

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RE: [Asterisk-Users] Last two digits getting cut off?

2005-07-21 Thread Kevin Walsh
Rob Engstrom [EMAIL PROTECTED] wrote:
 (Article auto-converted from unnecessary HTML to nice plain text.)

 We've just setup our [EMAIL PROTECTED] server, with our quad port card.  
 Everything
 works well so far. 
 
 One thing I notice is that when I leave the handset on the hook and dial
 a #, all is well.  If I pick up the phone and dial, it cuts off at 10
 digits, which is a problem if I need to dial 1+area+phone # (12 digits). 
 
 The phones are Poly Soundpoint IP 600's. I'm wondering if I've missed a
 config to allow more than 10 digits? 
 
The Cisco 7960 and Sipura devices have a dialplan that define when to
send the dialled number to the server.  If the SoundPoint has something
similar, and I expect that it does, then that would be a good place to
start looking.

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RE: [Asterisk-Users] RE: Business Edition

2005-07-21 Thread Kevin Walsh
Lee Howard [EMAIL PROTECTED] wrote:
 Kevin P. Fleming wrote:
  You seem to be neglecting the amount of work that Digium puts into the
  Asterisk (and related) products on an ongoing basis that is given to
  the community at no charge.
  
 So at least we agree, then, on what the reasoning is.  Digium feels that
 the community owes it to them.
 
I agree with that assessment.

It has been flippantly said, a number of times, that if you don't
like the situation then you can fork the project.  A major fork seems
(to me) to be pointless for one main reason (and a couple of lesser
reasons):

As I see it, anyone working on an Asterisk fork who had previously
signed the dangerous disclaimer (the perpetual one) could find their
changes to the fork rolled back into the Asterisk Binary Edition
without any further permission being required.

The perpetual agreement grants the owner a non-cancellable right
to use changes and/or enhancements made to the Asterisk codebase as
[the] owner sees fit.  As any Asterisk fork would, of course, be based
upon existing Asterisk code, the owner would have the automatic right
to take any code they wanted and backport it into the Asterisk Binary
Edition - as long as the contributor to the fork had previously signed
a perpetual disclaimer at some point in the past.

A fork wouldn't get very far without the support of at least some of
the regular contributors, all of whom have probably signed the
perpetual agreement.  For this reason, I believe that if a fork were
ever necessary, it would struggle to beat a distinct path away from
the Asterisk Binary Edition, which would be free to assimilate any
advances into its own codebase.

To mitigate the above, I believe that the perpetual disclaimer should
be modified to cover only a specific time period (i.e. one year from
the date of submission).  All contributions made within that time
period would be covered by the currently-valid agreement, and that
agreement could be renewed annually, if desired.  I don't see that
sort of change happening anytime soon because I believe that the
perpetual nature of the agreement is quite deliberate.

I think people sign agreements out of convenience, or pressure,
without reading carefully enough.  For instance, I wonder how many
people actually received their $1.00 (One Dollar) and other good and
valuable consideration when they signed their future options away.

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RE: [Asterisk-Users] Business Edition

2005-07-19 Thread Kevin Walsh
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
 On Monday 18 July 2005 23:50, Kevin Walsh wrote:
  I suspect that there is now less of an incentive to produce stable
  branches, and backport fixes to those branches from the development
  version, as this could possibly reduce the value of the closed version
  somewhat.  It could turn out that we eventually find the project in a
  permanent in development state, with no stable releases at all - just
  the CVS HEAD.  Once you start down that road, and rely upon revenue
  generated from closed source products, it's difficult to turn back.
 
 This is a typical slippery slope argument.  There has been no indication
 that this is what is occuring, nor that this is what will occur.  I
 personally suspect that at some point the 1.0.x series of Asterisk and
 ABE will end up being the same thing; a feature-frozen version of
 Asterisk.  ABE might have some kind of checksum or other
 authentication/verification wrapper to ensure that it's actually ABE and
 not almost ABE. 
 
Well, we can only hope so.  I suspect the opposite and, so far, all of
my negative fears and suspicions have come to pass.  It was very easy
to foresee that the disclaimer documents' sole purpose was to allow
for the future closing of the source.  This has been pointed out and
warned against for years, and now it is sad to see it starting to happen.
These are very worrying times for the future of the project.

  
   If you don't want or don't like ABE, don't use it.  Nobody is
   cramming it down your throat.
  
  That's not the point.
 
 What exactly is the point, then?   You don't like it because it's not
 free as in libre, but you don't propose any solid way for Digium to
 maintain its profitibility and grow in order to help support the
 community and make Asterisk grow.   I understand that you want Asterisk
 totally free but I don't see a way to do so, and I don't think you do,
 either. 

Digium can remain profitable by selling hardware and support.  I suspect
that if someone using the GPL version was to contact Digium and offer to
pay for support, they wouldn't be turned away for not using the Asterisk
Binary Edition.  The ABE allows limited support (whatever that means)
to be paid for in advance, possibly in the hope that a lot of people
won't actually use all of their allowance.  That's a good strategy, but
one that would still be possible with the GPL version.

I define free as libre, as you said.  This doesn't rule out the
possibility of charging for installation, support and customisation,
nor making money in other ways.  The Asterisk Binary Edition isn't
free in any sense of the word.

 
 You've presented a slippery slope argument and a strawman (disclaimers)
 and I am interested in finding out how you'd do it if it were your ship
 to steer. You're not a screaming zealot and I think this discussion's
 good for the list. 
 
If it were my project to steer, then there would be no disclaimers,
and therefore no possibility of a non-GPL release.  I believe the
project would move forward a lot quicker if everyone was given the
opportunity to offer code and patches under the terms of the GPL,
rather than simply relying upon a few committed developers who are
happy to donate their code to a single company to close and sell.

Don't get me wrong.  I use and recommend Asterisk, and I think the
software is great - a credit to the dedicated individuals who keep it
going.  I just don't like the one individual company which constantly
tries to claim all of the credit for what is clearly a community effort.

One piece of good news can be found here:

http://www.asterisk.org/index.php?menu=summer_of_code

The requirements say nothing about being asked to sign a disclaimer,
so perhaps either Google have views on this sort of practice, or people
will be quietly rejected, during the interview process, based upon their
willingness to have their source code closed.  I suspect the latter,
which would not make it good news after all.

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RE: [Asterisk-Users] Installation

2005-07-19 Thread Kevin Walsh
Aron Bereket [EMAIL PROTECTED] wrote:
 (Article auto-converted from unnecessary HTML to nice plain text.)

 I am a new user of Asterisk. When I downloaded it and was trying to
 compile it, i got the following error message after it run for a while.
 This error came after I run make. 
 
 bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c  
 ast_expr.y:110: unrecognized: %locations
 ast_expr.y:110:Skipping to next %
 ast_expr.y:141: invalid @-construct
 ast_expr.y:141: $. is invalid
 ast_expr.y:141: invalid @-construct

I suspect that you're running an ancient version of Bison.  If that's
the case then upgrade it and try again.

 
 Start your day with Yahoo! - make it your home page

Why?

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RE: [Asterisk-Users] Business Edition

2005-07-18 Thread Kevin Walsh
KRTorio [EMAIL PROTECTED] wrote:
 For $995+ (including support), a technical manual, and scripts, is it
 worth switching to the business edition?
 
Absolutely not.  If you find that you need $995 worth of support, some
time in the future, then I'm sure that you can obtain it from one of
several providers.  I don't think it's worth paying up-front for
something you probably won't need, but that's really for you to decide.

If you really want to pay $995 for a closed source product with some
features removed and license control added, then go for it.  It's
your money.

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RE: [Asterisk-Users] Business Edition

2005-07-18 Thread Kevin Walsh
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
 I dunno... people seem all up in arms about this but honestly I fail to
 see the problem.  Digium is doing what they can to make money and provide
 services while keeping Asterisk as free and openly developed as possible.

Services could be provided, and money could be made, without resorting
to selling closed source versions of the product.  Apparently, the
closed version consists of the contents of CVS HEAD, with various
changes made to increase reliability and decrease risk - according to
the FAQ.  It would be nice if the binary version's source was available
as a branch in CVS, but that probably doesn't fit into a closed source
business model very comfortably.

I suspect that there is now less of an incentive to produce stable
branches, and backport fixes to those branches from the development
version, as this could possibly reduce the value of the closed version
somewhat.  It could turn out that we eventually find the project in a
permanent in development state, with no stable releases at all - just
the CVS HEAD.  Once you start down that road, and rely upon revenue
generated from closed source products, it's difficult to turn back.


 I have (small amounts of) code contributed to Asterisk and I am working on
 more.  Digium and Asterisk have given me a lot of newfound freedom and
 flexibility and power in my phone system.  I appreciate that, and I don't
 feel that this dual-licensing or granting of a nonexclusive perpetual
 license to the bits and pieces of my code is too much to ask.  My bits
 and pieces would be worthless without the bits and pieces and chunks and
 slabs of code that others have provided, and it'd all be useless without
 the framework that Digium came out with. 
 
Asterisk would not be the product it is without the efforts of the
community who, it seems, have provided the majority of the source
code and support for the project.  Of course, Digium try their best
to not accept patches to their code unless they are accompanied with
a disclaimer.

According to the bug tracker (http://bugs.digium.com/main_page.php),
the disclaimers are insisted upon in order to keep copyright clean,
even though it has been pointed out, several times, that the agreements
have no effect on copyright at all.  The disclaimers exist to grant
Digium the right to close and sell your code.  If you're happy with that
then that's your choice to make.


 If you don't want or don't like ABE, don't use it.  Nobody is cramming it
 down your throat. 
 
That's not the point.

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RE: [Asterisk-Users] G.729 licensing - HardwareDevices rather than software

2005-07-18 Thread Kevin Walsh
trixter [EMAIL PROTECTED] wrote:
 On Mon, 2005-07-18 at 21:45 -0600, Tim Pushor wrote:
   Just gotta watch that you dont have two with  the same mac addr in
   some networks (some systems and network devices dont care enough
   others completly come unglued). 
   
  Yeah, like ethernet.
 
 let me clarify, on an ethernet network some systems and devices dont care
 others freak out. 
 
A better example would be the case where two machines are on different
Ethernet networks.  Perhaps two PBXs, in two separate offices, connected
via the Internet.

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RE: [Asterisk-Users] Zap timing device

2005-02-22 Thread Kevin Walsh
Umar Sear [EMAIL PROTECTED] wrote:
 I have been using asterisk for some time now. However I have never
 used it with any of the digium or compatable cards (Purely used for SIP).
 
 I understand that for using Meetme, I need to have a timing device,
 which could either be hardware or zrdummy etc (I am not using any right
 now). 

There's also zaprtc to consider.  See here:

http://www.voip-info.org/wiki-Asterisk+timer

 
 Can someone tell me if the timing device is needed for voicemail and
 other applications too?.
 
No - the timer is only required for conferences, IAX trunking and MOH.
There might be others I've missed, but Voicemail is not one of them.

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RE: [Asterisk-Users] Asterisk UK Community

2005-01-10 Thread Kevin Walsh
Ben Merrills [EMAIL PROTECTED] wrote:
 (Article auto-converted from unnecessary HTML to nice plain text.)

 To those who are/were interested J

 Ok, I’ve setup a mailman mailing list. I don’t so this often, so I’ve
 done a bad job of it, please let me know. Oh, and I know the SSL cert has
 expired… was self signed J Now I’ve apologised for my crapness, the URL
 to signup to the mailing list is:

If the list founder posts using HTML then that doesn't bode well.
I think I'll give it a miss and continue to use the main mail lists.

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RE: [Asterisk-Users] [Asterisk-Dev] RE: [Asterisk-biz]Asterisk training andcertification :: AstriconTraining

2004-12-21 Thread Kevin Walsh
 
 ...For Asterisk gurus, that believe that you can take the exam without
 attending the training, there will be exam oppurtunities setup in
 combination with Astricon conferences. When we update dCAP for future
 releases of Asterisk (1.1, 2.0), you will be able to upgrade your
 certification at Astricon... 
 
Great - so that's another $3000 every few months then.  I'll just
laugh at anyone who's stupid enough to show me their certificate.

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RE: [Asterisk-Users] asterisk server to asterisk server

2004-12-21 Thread Kevin Walsh
William Betts [EMAIL PROTECTED] wrote:
 what is the best way to have 2 asterisk servers communicate with each
 other?

Probably using IAX2.

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RE: [Asterisk-Users] soho usage

2004-12-21 Thread Kevin Walsh
Mathias Houngbo [EMAIL PROTECTED] wrote:
 i want to know if it is possible to use
 an analogic phone with a modem in asterisk ?
 
 i don't want to use a digium card or any FOX/FXS module !
 
 perhaps a modem card ??

You could consider a Sipura device.  The SPA-2000 comes with two FXS
ports, for two analogue phones, and the SPA-3000 comes with one FXS
and one FXO port.

Phone ports on modems usually just pass through to the phone line,
so I doubt that you could use anything like that to set up an analogue
phone as an Asterisk extension.

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RE: [Asterisk-Users] SOHO PBX using asterisk

2004-12-21 Thread Kevin Walsh
Christopher L. Wade [EMAIL PROTECTED] wrote:
 Oops, just noticed, for 5 (FIVE) phones plus using the analog line
 related to your ADSL line, you'd need a TDM40B and a TDM11B.
 
Or two Sipura SPA-2000s plus one SPA-3000.

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RE: [Asterisk-Users] SOHO PBX using asterisk

2004-12-21 Thread Kevin Walsh
Steve Wolfe [EMAIL PROTECTED] wrote:
 Kevin Walsh wrote:
  Or two Sipura SPA-2000s plus one SPA-3000.
  
 Can you tell us more about the sipura device?

A Sipura SPA-2000 provides two FXS ports for two analogue phones.
You can use a bunch of these devices if you need more ports.  Of
course, once you get to a certain number, a channel bank will be
cheaper.

A Sipura SPA-3000 provides one FXS port, as above, and one FXO
port for your phone line.

The configuration quoted above (2 x SPA-2000 and 1 x SPA-3000) will
provide 5 x FXS ports and 1 x FXO port, and should cost less than
$350 USD.

 
 Also if one is using sipura devices I would still be tempted to put in a
 real digium card -- or how well the the ztdummy  work for clocking?
 
If you need a hardware timing device for some reason then a cheap
X100P will do the trick.  The software timer modules should be ok,
on a Linux 2.6 kernel, for most uses.

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RE: [Asterisk-Users] modified prepaid application

2004-12-20 Thread Kevin Walsh
Adnan Ahmed [EMAIL PROTECTED] wrote:
 how can we integrate the modified-prepaid application with asterisk
 because when i compile the app_prepaid it gives bunch of errors.
 
I have no idea.  Perhaps you're out of disk space.

No?  Ah well - my mind reading powers seem to get weaker over long
distances.  Posting the error messages might help.

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RE: [Asterisk-Users] ATA Adaptor

2004-12-20 Thread Kevin Walsh
Jason Lane [EMAIL PROTECTED] wrote:
 I am new to asterisk and I am trying to get things set up so I can
 prove to the boss it works and get the budget to do a full
 implementation. Does anyone have an ata adaptor or an ip phone laying
 around they would be willing to sell me for around 30-50 dollars, I will
 need 2 of them. 

A Sipura SPA-2000 will cost you around $100 USD, brand new, and will
provide you with two FXS ports.  That should be enough for a quick
demonstration.  I don't know whether you were expecting each ATA to
provide one FXS port or two for your $30-50 USD budget.

If you don't get any offers from the subscribers of this mail list, you
could try eBay for a second-hand ATA device or two.

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RE: [Asterisk-Users] unsubscribe

2004-12-16 Thread Kevin Walsh
Bart de Wild [EMAIL PROTECTED] wrote:
 unsubscribe

No.

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RE: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOC NearYou!)

2004-12-14 Thread Kevin Walsh
Christopher Dobbs [EMAIL PROTECTED] wrote:
 My company has started development on a Ethernet based channel bank.
 
 Here are the (current) spec's
 - 10/100 Ethernet Port
 - Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired)
 - Serial Console
 - TDMoE
 - IAX2
 - EETP (A protocol that we have designed for IP Telephony)
 
How does your proprietary EETP protocol differ from the proprietary
IAX2?  I assume your protocol has support for trunking which, it seems,
is one of the main reasons why people use IAX2 instead of the SIP
standard.

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RE: [Asterisk-Users] list broken again?

2004-12-14 Thread Kevin Walsh
Greg - Cirelle Enterprises [EMAIL PROTECTED] wrote:
 It's been hours since I've seen a post from this list
 
 Must be broken again.
 
You determined that the mail list was broken and decided to alert
everyone by posting an article to the list?  Good thinking. :-)

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RE: [Asterisk-Users] Conjuring Kevin Walsh (was: four wildcards ina single pc)

2004-12-10 Thread Kevin Walsh
Gregory Junker [EMAIL PROTECTED] wrote:
  This is a good example for the newbies of the list as to why proper
  formatting and list ettiquitte is important. I made a mistake, it was
  easy enough for someone to come around behind me and correct the
  message. We all can make mistakes.
 
 Oh whatever, get off it already. Any minimally intelligent amoeba Would
 have understood the correction regardless of where it occurred in his
 post. This case is hardly the poster child for bottom-posting vs
 top-posting. Do you really want to start this nonsense up again?
 
He probably doesn't, but it seems that you do.  Why is that?

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RE: [Asterisk-Users] Asterisk from CVS

2004-12-10 Thread Kevin Walsh
Adi Linden [EMAIL PROTECTED] wrote:
 I admit that this might be some very basic question... How do I obtain
 Asterisk 1.0.3 from CVS? Does '-r v1-0' get me 1.0 or 1.0.3?
 
That will get you the very latest version of the 1.0 series, whatecver
that might be.  The version I have here shows me this:

Asterisk CVS-v1-0/2004-12-10/18:45:43/cursor-5, Copyright (C) 1999-2004 
Digium and others.

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RE: [Asterisk-Users] Door buzzer.

2004-12-06 Thread Kevin Walsh
Cian O'Sullivan [EMAIL PROTECTED] wrote:
 (Article auto-converted from unnecessary HTML to nice plain text.)
 
 They have a pizza box server as their asterisk server with a T1 card. No
 more slots, so if I want to use the existing infrastructure I will need
 to build a second server with an FXO port.  Kinda stupid having a second
 server just to open the door.  
 
If you need an FXO port and don't want to install a whole new server
then you could consider an external device, such as a Sipura SPA-3000.

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RE: [Asterisk-Users] Asterisk Hardware

2004-12-06 Thread Kevin Walsh
Walid Azab [EMAIL PROTECTED] wrote:
 (Article auto-converted from unnecessary HTML to nice plain text.)

 Can I start using Asterisk with a couple of SIP IP phones and Softphone
 software on users PCs only? I do not have any cards yet and will still
 have to wait until I order a card. 
 
Yes.

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RE: [Asterisk-Users] Users list.

2004-12-06 Thread Kevin Walsh
David Uzzell [EMAIL PROTECTED] wrote:
 Does this sudden rush of email mean we are all back online?
 
No.

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RE: [Asterisk-Users] G.729 algorithm?

2004-12-06 Thread Kevin Walsh
Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
 according to what I've found out this far, the G.729 patent seems not
 valid in a broad range of countries.
 
That is correct.

 
 does anyone know where I can find the algorithm?
 
G.729 consists of a lot of patents and a lot of algorithms.  A couple
of Google searches will scare up most of them.

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RE: [Asterisk-Users] G.729 algorithm?

2004-12-06 Thread Kevin Walsh
Robert Rozman [EMAIL PROTECTED] lazily top-posted:
 do you have info in what countries g.729 is not valid... ?
 
You could start with the whole of Europe and can also add the UK.
I'm sure there are lots of other countries who don't feel the need to
acknowledge US-based software and algorithm patents too.

This subject has been covered several zillion times in the mail list.
Google is your friend.

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RE: [Asterisk-Users] Gentoo and Asterisk - any experiences?

2004-11-30 Thread Kevin Walsh
Niels Chr. Sørensen [EMAIL PROTECTED] wrote:
 In constant search for optimization, a friend told us about his experience
 with Gentoo Linux-distro. He claimed that he doubled the performance of
 his server by changing to Gentoo from Debian.

 Does anyone have any experience with running Asterisk on a Gentoo linux?

I use Gentoo - both at home and at work.  You would see a vast
difference between an old 2.4-based distro and Gentoo running 2.6,
but a lot of that would be down to the scheduler in the 2.6 kernel.

If you compile from source then there's no doubt that you'll get a
speed boost.  I doubt that you'd double the performance without other
factors deserving most of the credit.

I do recommend Gentoo GNU/Linux with the 2.6 kernel.  Everything
just works (tm) without all the hoops you have to jump though with
other distros - such as Fedora.

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RE: [Asterisk-Users] Gentoo and Asterisk - any experiences?

2004-11-30 Thread Kevin Walsh
Patrick [EMAIL PROTECTED] wrote:
 On Tue, 2004-11-30 at 16:23 +, Kevin Walsh wrote:
  I do recommend Gentoo GNU/Linux with the 2.6 kernel.  Everything
  just works (tm) without all the hoops you have to jump though with
  other distros - such as Fedora.
 
 Not wanting to start the ol' distro wars again but I haven't come across
 any hoops I need to jump through while installing Fedora Core 2 or 3. My
 installs typically take 20 mins (start-end, including extensive
 individual package selection) which imho beats the lightyears it takes
 Gentoo to compile all the apps. Fedora Core 2 has worked fine for me
 running Asterisk for a long time now and I recommend that over Fedora
 Core 3 due to the introduction of udev which requires additional tweaks
 and too cutting edge versions of gcc for my taste.
 
Additional tweaks are classed as hoops to jump through in my view.
You're right about one thing though: We don't need another distro war.

By the way, a light year is a unit of distance - not time. :-)

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RE: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'

2004-11-24 Thread Kevin Walsh
Leandro Morgado [EMAIL PROTECTED] wrote:
 On Wed, 2004-11-24 at 13:53, Dave Cotton wrote:
  On Wed, 2004-11-24 at 13:45 +, Leandro Morgado wrote:
   Jose Hernandez wrote:
   
I installed TDM400P and X100P pci cards in a system running
mandrake 10.1 official, kernel 2.6.8.1-12mdksmp.
  
  
  This is not udev up to its tricks again?
 
 You are right! If his distro is using udev it might dynamically create
 the zap devices with the wrong set of permissions. I am not using udev
 (debian 2.4) but it makes sense that udev would allow you to specify the
 permissions to use when creating devices!
 
The following seems to work (udev permissions file):

# zaptel device
zap/*:asterisk:asterisk:644
zap/channel:asterisk:asterisk:660
zap/ctl:asterisk:asterisk:660
zap/pseudo:asterisk:asterisk:660
zap/timer:asterisk:asterisk:660

If you run Asterisk as root then the above is irrelevant, but nobody
does that - right?

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RE: [Asterisk-Users] Using IPKall and SIP with insecure=very

2004-11-22 Thread Kevin Walsh
Rob Emanuele [EMAIL PROTECTED] wrote:
 I've got one of those cool free incoming IPKall phone numbers from
 www.ipkall.com.  These numbers just connect to the SIP proxy of your
 choice, they default to Frreworld Dialup.  You can use them with your own
 sip proxy on asterisk.  My  config for this is below.
 
 The trouble I'm having is the incoming calls do not seem to hit the
 section in sip.conf for the call.  With sip debugging turned on I see the
 call come in and the message below is printed.
 
 If I put the exten route that I have in the ipkall-inbound section of
 extensions.conf (below) into the default section it works fine, but isn't
 neat and elegant. 
 
 How do I make incoming call from ipkall match a sip.conf section?
 
  From sip.conf:
 
 [3501]
 type=peer
 host=dynamic
 dtmfmode=rfc2833
 context=ipkall-inbound
 insecure=very
 nat=no

  From extensions.conf:

 [ipkall-inbound]
 exten = 3501,1,Goto(menu,s,1)
 
You'll probably find that there's no need to set up a specific
user for IPKall.  You were using type = peer, which would have been
wrong anyway.

In your [general] section, create a context = incoming-sip (or whatever
you want to call it) and then set up a matching context in extensions.conf.
Your extensions.conf context can then match your 3501 extension, along
with any other direct incoming SIP addresses you need.

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RE: [Asterisk-Users] Using IPKall and SIP with insecure=very

2004-11-22 Thread Kevin Walsh
Rob Emanuele [EMAIL PROTECTED] wrote:
 Kevin Walsh wrote:
  You'll probably find that there's no need to set up a specific
  user for IPKall.  You were using type = peer, which would have been
  wrong anyway. 
  
  In your [general] section, create a context = incoming-sip (or
  whatever you want to call it) and then set up a matching context in
  extensions.conf. Your extensions.conf context can then match your 3501
  extension, along with any other direct incoming SIP addresses you need.
  
 What if I wanted to create different incoming-sip contexts depending on
 the service being used or number being called?  For example sip calls
 coming from ipkall goto one context that presents a menu, but another sip
 call coming from one of the free German services provides a different
 menu and in German. 
 
Sip.conf is quite flexible and can be set up in lots of different ways.
If you had multiple accounts with multiple providers then you'd want
to set up separate user/peer sections for each provider, and could
easily set up each section to make use of its own specific context.

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RE: [Asterisk-Users] UK available SIP phone?

2004-11-21 Thread Kevin Walsh
Bill Seddon [EMAIL PROTECTED] wrote:
 I use Asterisk at home and have bought a couple of HandyTones ATAs.  The
 DECT phones are plugged in and work really well.  The ATAs are £56 from
 Goods2World (though the additional one I've just bought didn't work and is
 being returned) and about the same from VoIPTalk (who are out of stock
 currently)

 The only downside to the ATA is that I've not yet worked out how to have
 CallerID displayed on the DECT phones.

A Sipura SPA-2000 would be cheaper than two HandyTone ATAs, and Sipura
support the Caller ID name/number display on DECT phones without any
problems.

You can get one from CallUK for £85+VAT (http://www.calluk.com/sipura/).

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RE: [Asterisk-Users] i swtiched to digest

2004-11-19 Thread Kevin Walsh
FuturaHost.Com Lists [EMAIL PROTECTED] wrote:
 I believe the list is so big that many of us are loosing some
 interesting threads. May be the admins can split the users list in some
 more specific sub-lists, and the people who wants to receive all the
 messages can subscribe to the sublists, or have a digest for someones,
 etc. 
 
You'll find that many people will want to be subscribed to all of the
mail lists - just in case something interesting is said or asked.

You'll also find that some Muppets will post their questions to multiple
lists, instead of finding a specific list to use, or will judge that
their question/comment has relevance in multiple lists.  For instance,
how many Asterisk veterans are likely to hang out on asterisk-newbies so
that they can answer the same FAQ question every ten minutes, and how
many Asterisk users are going to post their questions to the newbie
list when they find that there are no experts there?  People already
post and/or duplicate end-user questions to the developers' list as
it is.

Splitting up the mail lists will most likely result in more bandwidth
use for most people - not less.

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RE: [Asterisk-Users] rtp codec error

2004-11-19 Thread Kevin Walsh
Daniel Eboa [EMAIL PROTECTED] wrote:

 Hello all,

And that's as far as I read.

You should try re-posting your question without the HTML, without the
bold multi-coloured text and without the pointless, out-of-focus
graphics.

Having done that, perhaps someone will be inclined to read your article
and might even be able to help.

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RE: [Asterisk-Users] i swtiched to digest

2004-11-19 Thread Kevin Walsh
Chris TenHarmsel [EMAIL PROTECTED] wrote:
 I've ammased 12Mb of email in a weekthis list does generate a LOT
 of traffic that it gets the point for me that it's really hard to wade
 through it all. 
 
Just do what I do:

1. Scan the subject list and delete anything that doesn't appear
   to be of interest.

2. Delete any messages that have lazy, top-posted followups.
   A rule that marks articles that include -Original Message-
   in the text will find a whole load of these, and will make life
   a lot easier.  Others will be found and deleted as and when they
   are opened, assuming they passed rule #1, of course.

3. Delete all HTML articles.  I don't have a rule for these at the
   moment, so I usually just delete them as and when I find them.

Following the above, you'll find that you're left with only 10% or so of
your original mailbox size.  This 10% usually results in the articles
that are worth reading.

Of course, everyone will have their own way of trimming public mail
lists to a manageable size.  This is mine.

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RE: [Asterisk-Users] Re: Re: i swtiched to digest

2004-11-19 Thread Kevin Walsh
FuturaHost.Com Lists [EMAIL PROTECTED] wrote:
 Anyone wanting to split the list in more parts?
 
 Yes and no would suffice, so we can close this without a talk long a
 year, and without someones forcing their point of view to others.
 
No.

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RE: [Asterisk-Users] Lobotomized Sipura SPA-3000 configurationneeded

2004-11-18 Thread Kevin Walsh
Gregory Junker [EMAIL PROTECTED] wrote:
 One problem is that the SPA3K only uses two-stage dialing on the FXO --
 VoIP2 path, which means any time someone calls the phone system and gets
 forwarded to a select SPA3K extension, they hear a dial tone. As far as
 I can tell, there is no way to disable that. You can have it execute a
 particular dialplan in the SPA3K but the caller gets to hear the digits
 as they are dialed into Asterisk.
 
You didn't say which firmware version you have, but I suspect that it's
out of date.  A quick update should fix that.

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RE: [Asterisk-Users] SS7 for *

2004-11-18 Thread Kevin Walsh
Matthew Crocker [EMAIL PROTECTED] wrote:
  Here's a question: if the author has purchased a commercial license to
  use Asterisk, and I get binary modules from him, I can still use them
  with my CVS-based Asterisk, right?
 
 You may be able to do that.  You could always run a couple Asterisk
 boxes, run IAX2 between them and leave the commercial stuff with SS7 in
 stock (aka supported) configuration.
 
I just avoid people who think it's ok to create proprietary extensions
to free software.  People like that should be ashamed of themselves,
as it's just an insult to the people who have freely contributed to
the project.

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RE: [Asterisk-Users] Re: Top posting

2004-11-17 Thread Kevin Walsh
Matt Riddell [EMAIL PROTECTED] wrote:
 Stephen R. Besch wrote:
  This all reminds me so much of Jonathan Swifts bit about the BigEndians
  and the LittleEndians (referring to which is the 'correct' end to open a
  soft boiled egg) in Gulliver's travels.
 
 But that's simple, surely you should put the big end of the egg into the
 egg cup and open from the tapered end, so as to avoid the egg falling
 over and losing its contents! :-)
 
Of course, some people wouldn't spot that one end had already been
opened and would proceed to open the other end as well.  Unfortunately,
these idiots usually end up making a complete pig's breakfast of the
whole thing while irritating the more civilised diners. :-)

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RE: [Asterisk-Users] The Apperiant Death of IAXtel

2004-11-17 Thread Kevin Walsh
Christopher Dobbs [EMAIL PROTECTED] wrote:
 Given that IAXtel has not been responding for some time, I am willing to
 setup accounts for thoes who want to have that kind of functionallity.
 If you are interested, send me an email with your requested username and
 password, and i will send you your account information.
 
Given that IAXtel has not been responding for some time, what use is
a username and password?

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RE: [Asterisk-Users] Re: Top posting

2004-11-14 Thread Kevin Walsh
Paul Fielding [EMAIL PROTECTED] lazily top-posted:
 Whatever.  I find it frankly more annoying to have people bottom post.  I
 use Outlook Express for my mail (as do millions of others), and the way OE
 formats it's mail lends itself to top posting.

As you seem to find it difficult to move the cursor on your own,
perhaps this utility will help:

http://home.in.tum.de/~jain/software/oe-quotefix/

You could install it to fix your broken mail reader - if it's not too
much effort.


 When you bottom post, I
 need to scroll way down the message to see your response

The effort involved is clearly too much for you to handle.  Are you
really that lazy?


 If I want to see the source
 message *then* I'll scroll down, but chances are I've already been reading
 the thread so this isn't necessary.
 
Your laziness will make life difficult for people who find your followups
in a future Google search.  Just because you've read the entire thread,
doesn't mean that someone else will have done the same next year.  Then
again, the chance of you posting useful information for someone to find
in Google does seem to be a bit remote.


 just my 2 cents

That might be all your time is worth.  Others get paid a little more
than that.

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RE: [Asterisk-Users] Re: getting callerid from spa3k to asterisk

2004-11-14 Thread Kevin Walsh
Randy Bush [EMAIL PROTECTED] wrote:
  if i have two sip contexts for my spa3k, on inbound and
  one outbound, e.g.
  
  [spa3k-in]
  type=friend
  host=dynamic
  port=5061
  auth=md5
  secret=pfui
  qualify=1000
  canreinvite=yes
  context=ext-in42
  
  [spa3k-out]
  type=peer
  auth=md5
  secret=pfui
  username=outpass
  fromuser=outpass
  host=spa3k.bogus.com
  port=5061
  nat=no
  canreinvite=yes
  context=ext-in42
  
  and the spa3k's PSTN / Subscriber Information / User ID: = spack-in,
  
  the incoming connection from spa3k to * is being routed to the
  spa3k-out context, not the spa3-in context.  see appended.
  
  i suspect this is a bug in * 1.0.1.
 
 i found the problem, or at least a work-around.
 
 if i reverse the order of the above two sip contexts, the incoming
 call is properly routed to the spa3k-in sip context as opposed to the
 wrong one, spa3k-out. 
 
 my guess is that * is traversing a list and taking the first
 context which has the ip address and port it wants without
 checking the context name against the name which was received
 over the wire.  so it depends on what order the contexts are inserted in
 the list. 
 
Yes, but it's not a bug. :-)

You have type=friend and type=peer.  Change friend to user for the
incoming definition.

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RE: [Asterisk-Users] Re: Top posting

2004-11-12 Thread Kevin Walsh
Bill Seddon [EMAIL PROTECTED] lazily top-posted:
 No, he didn't.  But I'd guess that readers of his email who are, perhaps,
 less sophisticated users of email lists will have been concerned about the
 attitude he projected and be unnecessarily anxious about posting questions
 in the wrong way.  Since there is no wrong way I just wanted to try to
 project a more flexible attitude.
 
Perhaps you should adopt a more flexible attitude and learn to follow
up properly.  There is no excuse at all for lazily top-posting.

There is a wrong way, and your method of posting demonstrates it clearly.
Please locate your mouse, arrow keys or whatever you use to move your
cursor around, and make an effort to post your followups in context.

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RE: [Asterisk-Users] Motherboard whitelist (was Echo - UK Impedanceproblem with X100P?)

2004-11-12 Thread Kevin Walsh
 On November 12, 2004 09:46 am, Rich Adamson wrote:
  As far as the motherboard issue, its not just the digium products that
  have an issue. If you know of someone that is heavy into using audio
  applications (eg, song writers, midi stuff), they have known about the
  pci / interrupt latency issues on certain motherboards for a lng
  time. Wouldn't doubt those folks maintain a list of what's reasonable
  verses unacceptable.
 
 You bring up an excellent point -- Can we use the blacklists and
 whitelists the audio folk have to help the asterisk community?  They'll
 have all hte same issues -- shared interrupts, craptastic PCI chipsets,
 etc...  sounds like a great idea to me. 
 
Why not just get a Sipura SPA-3000 and side-step the motherboard vs. TDM
card compatibility issue altogether?

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RE: [Asterisk-Users] Re: Top posting

2004-11-12 Thread Kevin Walsh
Richard Cook [EMAIL PROTECTED] lazily top-posted:
 Must be nice to have time and money to worry about someone's posting
 method. Amazing. 
 
Time is money, and I will waste neither trying to work out what it is
you just followed up to.  Perhaps if you learned to post in context...

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RE: [Asterisk-Users] UK BT Caller ID, X100P and Asterisk v1

2004-11-09 Thread Kevin Walsh
StrUK [EMAIL PROTECTED] wrote:
 Having one
 card with FXO and FXS modules on and consigning the X100P to the heap is
 a real goal - I need IRQs! 
 
Get a Sipura SPA-3000 and forget all about IRQs.

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RE: [Asterisk-Users] X100P CLONES again

2004-11-09 Thread Kevin Walsh
Seth Remington [EMAIL PROTECTED] wrote:
 On Tue, 2004-11-09 at 15:22, [EMAIL PROTECTED] wrote:
  What model of modems can be use as X100P?
  
  I can get a Motorola 62802-52, did anybody ever try it?
  
  I must buy some of the clones because in my country nobody sells
  anything of voip yet. 
  
 Check the Generic heading:
 http://www.asterisk.org/index.php?menu=hardware
 
To see the product prices, call our Sales Department now at 999 and
ask for the password. - Ha!

If they're too embarrassed to display their prices then just avoid them.
Why should you have to put in extra work just because they couldn't be
bothered to put a proper e-commerce system in place.

You'll find lots of alternative X100P products if you look around in
Google and eBay.  They are all clones of the same design - even the
Digium product.  Some are just more expensive than others.

You should also check whether the Sipura SPA-3000 is technically
compatible with the PSTN system used in your country.

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RE: [Asterisk-Users] Digium Generic Boards - Low Prices / High Quality.

2004-11-09 Thread Kevin Walsh
Richard Moore [EMAIL PROTECTED] wrote:
 As announced before we're starting selling only E100P based
 boards. T100P boards and other digium-like products will
 be available in 20-30 days.
 
 All products are being tested for our engineers. We're developing
 our website with all products and we'll include a web shopping for global
 sales. 
 
 At this momment , i'm at Brazil negotiating a distribution channel for
 latin america. 
 
 500 units were sold only today. (E100P)
 
 So, please wait for our products to be released.!
 
 don't waste money :)
 
Although your products are probably of interest to a large number
of Asterisk users, you'll probably want to move your adverts onto the
asterisk-biz mail list before people start whining.

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RE: [Asterisk-Users] E100P - Generic (Clone) - :)

2004-11-09 Thread Kevin Walsh
Tom [EMAIL PROTECTED] wrote:
 So Richard, since you are a Asterisk Engineer, will you take over
 Asterisk software development if Mark Spencer cannot sell hardware to
 stay interested? 
 
You probably failed to notice the Asterisk community.  Asterisk is
larger than one man or company.  Some Asterisk developers even post to
the mail lists and answer questions etc.

Competition is a good thing.

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RE: [Asterisk-Users] X100P CLONES again

2004-11-09 Thread Kevin Walsh
Richard Lyman [EMAIL PROTECTED] wrote:
 Kevin Walsh wrote:
  Seth Remington [EMAIL PROTECTED] wrote:
   Check the Generic heading:
   http://www.asterisk.org/index.php?menu=hardware
  
  To see the product prices, call our Sales Department now at 999 and
  ask for the password. - Ha!
  
  If they're too embarrassed to display their prices then just avoid them.
  Why should you have to put in extra work just because they couldn't be
  bothered to put a proper e-commerce system in place.
  
  You'll find lots of alternative X100P products if you look around in
  Google and eBay.  They are all clones of the same design - even the
  Digium product.  Some are just more expensive than others.
  
  You should also check whether the Sipura SPA-3000 is technically
  compatible with the PSTN system used in your country.
  
 maybe it's you who should be embarrassed.  you see, they are a
 distributor, and therefore you don't get an account without a resale
 license. 
 
As all of the other links on that asterisk.org page are for retail
sales, perhaps the so-called generic link should point at a retail
website.  As I said, a quick look on Google will reveal plenty of them.

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RE: [Asterisk-Users] UK BT Caller ID, X100P and Asterisk v1

2004-11-09 Thread Kevin Walsh
Richard Hamnett [EMAIL PROTECTED] wrote:
  Get a Sipura SPA-3000 and forget all about IRQs.
  
 How about just enable kernel APIC support and forget all about IRQs!!
 
I thought everyone was using APIC now anyway.  Why would someone not
have it enabled?

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RE: [Asterisk-Users] re: CallerID for the UK

2004-11-08 Thread Kevin Walsh
Charles Osstyn [EMAIL PROTECTED] wrote:
 (Article auto-converted from unnecessary HTML to nice plain text.)

 
 Hi all, I am too new with Linux, to really experiment with the callerid.
 I know the problem is due to BT using a different technical platform. 

 So as too using Perl or any other scripts and a modem to create the work
 around required to get the callerid to work UK I need some help in this
 field.  
 
 Anyone, got a step-by-step guide, how to add this to a working setup?

If you're using an X100P then I have some patches that will give you
UK (BT) Caller*ID without having to resort to the use of modems and
scripts.  Email me off-list if you want the patch files.

 
 This would solve my last technical issue. Many thanks in advance also any
 one got any RPM's for a GUI which can be used to setup SIP and IAX
 account and configure the dial plan through a nice web interface. 

There are several web-based interfaces out there.  Whether you consider
any of them to be nice is up to you. :-)

 
 [snip: 40 line signature, unnecessary attachments and adverts]


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RE: [Asterisk-Users] Linux and Windows

2004-11-02 Thread Kevin Walsh
Jay Milk [EMAIL PROTECTED] lazily top-posted:
 I'm not sure why I'm even discussing the benefits of one operating
 system over another
 
I don't know why you bother posting at all.

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