[Asterisk-Users] SQL Updater Down!!!

[Chandra]

hi,

I am trying to install ASTGUICLIENT and when i run the
AST_WINphoneAPP_0.8.pl it opens a window  VICI Phone App -0.8 but i am
getting SQL Updater Down Mesasge. How can i solve this?


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Re: [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino

[Eric Wieling]

On Thu, 2004-01-01 at 12:25, JR Richardson wrote:
> That did it.  I ran that export command you suggested, then launched *,
> everything worked fine.  I'm still looking for info on what that command
> actually does.  Can you shed some light please?

It's in the RedHat RELEASE NOTES.



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RE: [Asterisk-Users] New to asterisk? RUN... don't walk.

[Girish Gopinath]

Excellent!!! Well Said, JR...

From: "JR Richardson" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] New to asterisk?  RUN... don't walk.
Date: Thu, 1 Jan 2004 10:11:36 -0600
Piping in 2 cents,

This is a great example of the Internet, Fast Food generation, showing 
their
appreciation for all the magic that happens in the labs, hearts and minds 
of
the courageous, hard working, dedicated and motivated group of people truly
interested and guided to accomplish greatness.

This platform for learning is one of the best tools in existence to come to
a finite understanding of VoIP and legacy telephony with the versatility to
expand beyond and develop originality in the field of telecommunications
excellence, product development.  Learn it, understand it, appreciate it,
then take it past where you found it and if you're capable contribute, if
not, enjoy it.  But always, always maintain respect for those who created 
it
and continue to refine it.

Learning is intrinsically human, and in this world of Industry ("There is 
no
substitution for knowledge." [Edward Deming]).  Find your inner child,
re-capture and embrace what God has given you, the ability to learn.  It
will require you to put down the remote control, get off the couch and
decrease your apparently frequent visits to McDonalds.  Search and find the
knowledge which you seek to ultimately fulfill your destiny; build an
Asterisk Server that works.

Hell, we all did.

JR





> Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST)
> From: Me <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] New to asterisk?  RUN... don't walk.
> Reply-To: [EMAIL PROTECTED]
>
> As a newcomer to Asterisk, you will not be welcomed
> with open arms.  First, you will find almost no
> documentation on it's features.  Second, if you try to
> ask questions, you will be flamed and pointed to
> worthless how-tos and 'the wiki'.  These worthless
> documents can only be useful for explaining how things
> work to those already in-the-know.  Lastly, Asterisk
> is so bug ridden, expect frequent segmentation faults.
>  With a community so 'anti-n00b', don't expect your
> problems to be fixed anytime soon.
>
> RUN!!! Don't walk... away from Aterisk.
>


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Send DD, pay no commission. 
http://server1.msn.co.in/msnleads/suvidha/dec03.asp?type=hottag Click here.

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Re: [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino

[James H. Cloos Jr.]

> "JR" == JR Richardson <[EMAIL PROTECTED]> writes:

JR> I ran that export command you
JR> suggested, then launched *, everything worked fine.  I'm still
JR> looking for info on what that command actually does.  Can you shed
JR> some light please?

Exporting LD_ASSUME_KERNEL=2.4.1 tells libc to use the old-style
'int 80' method of doing syscalls to the kernel, as well as the
old-style type of thread support.  RH9 and 2.6 kernels support
newer, faster methods of syscalls and threads on amd64 and recent
ia32 cpus.  The need to assume an earlier kernel version indicates
that some part of * or a lib it (or one of its modules) is linked
to breaks when using the newer routines.

Eventually such bugs should be eradicated and LD_ASSUME_KERNEL will
not be required.  (Eg, my gentoo laptop only supports nptl threads
and I have no problems running * there.)

-JimC


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Re: [Asterisk-Users] sound driver advise needed

[Steven Critchfield]

On Thu, 2004-01-01 at 22:48, Steve Murphy wrote:
> Hello--
> 
> How do I twiddle the sound drivers?
> 
> I'm not that experienced with kernel twiddling and driver loading. I
> have Redhat 9. My previous attempts to play with the kernel and load
> extra drivers always ended with a new kernel that wouldn't boot.
> 
> I know asterisk doesn't like to play with sound interfaces that aren't
> full duplex. So, when I built the
> system, I saw the MSI motherboard didn't explicitly say the sound
> chips were full duplex, so I threw
> in a Soundblaster PCI card, that DID advertise full duplex.
> 
> Trouble is, asterisk only sees the brain-dead interface. How do I
> exorcise it from the kernel, or at least make the SB the
> first-priority one?  rmmod didn't seem to do anything. Playing with
> the Redhat sound  card detection stuff was useless. I've googled
> around the internet, looking for tidbits, but nothing seems
> applicable. RedHat 8 Bible wasn't very helpful.  Not much in the
> kernel source dirs, either, nor in the source for the sound drivers.
> 
> Anybody have some experience with this sort of thing? It'd be neat to
> put together an announcement functionality.
> 
> Any advise? Many thanks,

Go into your bios and turn off the internal sound card. Thats how you
take care of it.

Oddly enough, the MSI should be full duplex, there isn't any reason for
someone to be so cheap as to not include decent chips anymore. 

You also should probably think about whether or not you really want to
use the console in asterisk. Sound card usage is pretty easy in VoIP
softphones, and you can disable the sound cards use in asterisk. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] sound driver advise needed

[Steve Murphy]

Hello--

How do I twiddle the sound drivers?

I'm not that experienced with kernel twiddling and driver loading. I have Redhat 9. My previous attempts to play with the kernel and load extra drivers always ended with a new kernel that wouldn't boot.

I know asterisk doesn't like to play with sound interfaces that aren't full duplex. So, when I built the
system, I saw the MSI motherboard didn't explicitly say the sound chips were full duplex, so I threw
in a Soundblaster PCI card, that DID advertise full duplex.

Trouble is, asterisk only sees the brain-dead interface. How do I exorcise it from the kernel, or at least make the SB the first-priority one?  rmmod didn't seem to do anything. Playing with the Redhat sound  card detection stuff was useless. I've googled around the internet, looking for tidbits, but nothing seems applicable. RedHat 8 Bible wasn't very helpful.  Not much in the kernel source dirs, either, nor in the source for the sound drivers.

Anybody have some experience with this sort of thing? It'd be neat to put together an announcement functionality.

Any advise? Many thanks,

murf

[Asterisk-Users] Re: How to load the driver of TDM400P card!

[Steve Murphy]

Quan--

> 
> I have just bought the X100P and TDM400P cards to install on my
> computer to implement the PBX. I also downloaded the newest softwares
> asterisk_ver0.5.0, libpri_ver0.4.0, and zaptel_ver0.7.0) to install on
> my computer (Red Hat Linux 8.0). All packages are compiled well. When
> I use "modprobe" to load drivers (modprobe zaptel, modprobe wcfxo,
> modprobe 
> 
> wcfxs), the first two (zaptel, wcfxo) are successful, but the last
> (wcfxs) is failed. This is the type of error:

I've been thru the same ordeal, almost exactly. A few things to
remember:

1. The order of modprobing should follow the channel declarations in the
/etc/zaptel.conf file. In mine, I have:

fxsks=1,2
fxoks=3-6

So, I do the modprobes in this order

modprobe wcfxo
modprobe wcfxs
ztcfg

Remember that fxs and fxo have inverse interface/signaling
relationships, and that the zapata.conf file is declaring the signaling,
and the drivers concern the hardware interface.

Oh, and one other thing: I don't do a modprobe for zaptel; I think 
the ztcfg takes care of that, if it isn't pulled in via dependencies.

Now, the first modprobe gives errors because of the other channel
declarations, I think. The second modprobe generates no messages.

So, to sum it up: if you have problems, reverse the order of the
modprobes.

Also, pay attention to the results of 'cat /proc/interrupts', you want
your wcfxs and wcfxo devices to have their own interrupt slots, with the
slot numbers < 15, I think. Try to get them their own slots. You can
usually play around with the BIOS on the system, the plug-n-play
settings, I think. You may also have to worry about which board is
plugged into which slot, as some BIOS setups will tie the interrupt slot
to the PCI card slot on the motherboard. You may have to turn off a
bunch of stuff in the BIOS that wants interrupt slots, like USB stuff,
IO interfaces, etc, to cut down the competition for interrupt attention.


> /lib/modules/2.4.18-14/misc/wcfxs.o: init_module: No such device
> 
> > Hint: insmod errors can be caused by incorrect module parameters, 
> 
> > including invalid IO or IRQ parameters.
> 
> > You may find more information in syslog or the output from dmesg
> 
> > /lib/modules/2.4.18-14/misc/wcfxs.o: insmod 
> 
> > /lib/modules/2.4.18-14/misc/wcfxs.o failed
> 
> > /lib/modules/2.4.18-14/misc/wcfxs.o: insmod wcfxs failed
> 
> > ..."
> 
>  
> 
> For the X100P card, after loading its driver, I tested its features
> and they are OK! But I can not load the driver of TDM400P card.
> 
This makes sense. The x100p driver, wcfxo, is loaded first, but declared
second in zapata.conf. So, it gets its slots, but at the price of the
wcfxs slots.

> 
> I also find in the documents of Digium, but I only find the the way to
> 
> loaddrivers for T400P/E400P (modprobe tor2), or T100P/E100P (modprobe 
> 
> wct1xxp). I try these options for TDM400P, but it does not work.


Uh, options for the T400 stuff most certainly won't apply to the  TDM
interface.

murf


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Re: [Asterisk-Users] How to load the driver of TDM400P card!

[Tilghman Lesher]

On Thursday 01 January 2004 21:44, Quan Le Trung wrote:
> Hi!
>
> I have just bought the X100P and TDM400P cards to install on my
> computer to implement the PBX. I also downloaded the newest softwares
> asterisk_ver0.5.0, libpri_ver0.4.0, and zaptel_ver0.7.0) to install
> on my computer (Red Hat Linux 8.0). All packages are compiled well.
> When I use "modprobe" to load drivers (modprobe zaptel, modprobe
> wcfxo, modprobe
> wcfxs), the first two (zaptel, wcfxo) are successful, but the last
>
> (wcfxs) is failed. This is the type of error:
> > "...
> > /lib/modules/2.4.18-14/misc/wcfxs.o: init_module: No such device
> > Hint: insmod errors can be caused by incorrect module parameters,
> > including invalid IO or IRQ parameters.
> > You may find more information in syslog or the output from dmesg
> > /lib/modules/2.4.18-14/misc/wcfxs.o: insmod
> > /lib/modules/2.4.18-14/misc/wcfxs.o failed
> > /lib/modules/2.4.18-14/misc/wcfxs.o: insmod wcfxs failed
> > ..."

Is this the "type of error", or is this the exact error?  It does
matter.  The first line tells you the problem:  no such device.
The driver cannot detect the device on the PCI bus, which could
mean a couple different things.  It could mean that you don't have
the Molex 4-pin connector connected to the card, which means the
card isn't powered.  It could also mean that the card is faulty or is
not seated correctly or that there's an odd conflict between your
motherboard and the card that does not allow the card to be recognized.

In any case, it's a hardware problem.  If the molex power connector
is not connected, get that plugged in.  If it's plugged in correctly,
try a different slot, or even a different motherboard.

-Tilghman

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[Asterisk-Users] How to load the driver of TDM400P card!

[Quan Le Trung]

Hi!

 

I
have just bought the X100P and TDM400P cards to install on my computer to
implement the PBX. I also downloaded the newest softwares asterisk_ver0.5.0,
libpri_ver0.4.0, and zaptel_ver0.7.0) to install on my computer (Red Hat Linux
8.0). All packages are compiled well. When I use "modprobe" to load
drivers (modprobe zaptel, modprobe wcfxo, modprobe 

wcfxs),
the first two (zaptel, wcfxo) are successful, but the last (wcfxs) is failed.
This is the type of error:

 

>
"...

>
/lib/modules/2.4.18-14/misc/wcfxs.o: init_module: No such device

>
Hint: insmod errors can be caused by incorrect module parameters, 

>
including invalid IO or IRQ parameters.

>
You may find more information in syslog or the output from dmesg

>
/lib/modules/2.4.18-14/misc/wcfxs.o: insmod 

>
/lib/modules/2.4.18-14/misc/wcfxs.o failed

>
/lib/modules/2.4.18-14/misc/wcfxs.o: insmod wcfxs failed

>
..."

 

For
the X100P card, after loading its driver, I tested its features and they are
OK! But I can not load the driver of TDM400P card.

 

I
would like to receive your instructions and look forward to your response soon!

 

Yours
sincerely,

Quan
Le T.

 

P.S

I
also find in the documents of Digium, but I only find the the way to 

load
drivers for T400P/E400P (modprobe tor2), or T100P/E100P (modprobe 

wct1xxp).
I try these options for TDM400P, but it does not work.

 

 

[Asterisk-Users] Prediction for 2004

[Philipp von Klitzing]

Taken from http://voxilla.com/Article40.phtml (see also slashdot):

Here are our Top-Ten predictions for VoIP in ’04:
[...]
3. Asterisk hits it big. Committed users of the terrific open source PBX-
plus software develop easy-to-follow installation and configuration
menus. Asterisk  installations by small businesses and SOHO and home
users will soar. Asterisk-to-asterisk networks, bypassing the Bells and
even VoIP providers, begin to take shape.


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Re: [Asterisk-Users] Video

[jna]

I Guess I should have specified I am looking for windows clients. I seen the
gnophone on the * site but that was the only reference for anything video
and that's only Linux. I figured there HAS to be something out there just
not well documented and the search engines did not results in anything
definitive.

Thanks,
John


> On Thu, 1 Jan 2004, Olle E. Johansson wrote:
>
> > [EMAIL PROTECTED] wrote:
> > > What is the best software video client that is compatible with * ?
> > To add a follow-up?
> > Which channels support video and how?
> > I know that there's support in H.323 and SIP. Anything else?
>
> IAX (and IAX2), of course.
> gnophone supports this, for example. Though gnophone isn't exactly
> well-maintained, these days...
>
> > There's nothing documented here on the Wiki, a black hole that needs to
disappear ;-)
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Re: [Asterisk-Users] after hours - is this logic ok ?

[Lance Arbuckle]

Steven Critchfield wrote:
> 
> 
> First, one can not be flamed to death on a mailing list.
> 

well... back in the early days of the internet that was quite true
because there was a Small Company that made excellent FLAME suits.  The
suits were were made of top quality materials and were expensive to
manufacture and but the company charged a fair price and the suits were
embraced by the customers.  Then, along came another much larger
company, who even though they wern't in the FLAME suit business, decided
that expansion into that market would benefit their existing customer
base. So Big Company bought a FLAME suit from Small Company and proceded
to Embrace and Extend the flame suit technology for the benefit of all
mankind.  Some months later, Big Company declared the BC_FLAME Suit1.0
finished and started bundeling it with all their existing products at no
additional charge.  Even though the BC_FLAME Suit 1.0 had some problems
keeping flames from penetrating the "when nature calls" flaps and it was
discovered that the BC_FLAME Suit 1.0 was manufactured with materials
derived from the gypsie moth's favorite food, poor old Small Company
still found it's sales steadily shrinking.  Shortly after, Small Company
collapased leaving the only source of flame suits our friends at Big
Company.  Over the years, the BC_FLAME Suit has gone through several
enhancements and is now on it's 5.0 release but I still don't trust it
like my good old original FLAME suit from Small Company.  :)


-- 
  .~.
  /V\LCA
 // \\   
/(   )\  
 ^'~'^
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Re: [Asterisk-Users] after hours - is this logic ok ?

[Steven Critchfield]

On Thu, 2004-01-01 at 12:45, Lance Arbuckle wrote:
> For example, if I want to implement feature xyz, currently I go googling
> and read, read, read.  I may come up with two or three sample contexts
> that accomplish the same thing via different means.  All I can do now is
> pick one of the examples and modify to my needs without any regard to
> the pros and cons of each example.  Now, I *try* to post intelligently
> phrased questions to this list but after seeing some of the threads the
> past few days, I'm hesitant to post the "how do I implement this?"
> questions for fear of being flamed to death :( or getting the standard
> "Google is your friend" responses.  What would be a great help to me and
> possible other beginners, is whenever a question is asked of the list,
> responses include a brief explanation of the "why" for the particular
> answer.  Or, if there is another resource that talks in generalities
> about how to do thing in Asterisk, please, I beg the list, to tell me
> where it is :)

First, one can not be flamed to death on a mailing list. 

Secondly, it sounds like you have questions that would ask for
clarification of what has been read in the archives. This actually is a
good thing. If you read the ESR page about how to ask a question, it
tells you how to effectively ask a question is such a way you are likely
to get the answers you want. You already seem to be well on your way to
understanding how to ask these questions. Basically it boils down to
showing that you have done some research yourself, explain the problem
you are trying to solve, explain either how you are attempting to solve
it, or in your case the two methods you think will solve your problem.
At this point you have documented your effort, and proved you are trying
to help yourself.  This helps as it gives plenty of information about
where you are currently in your understanding of asterisk, so answers
given can be tailored a bit to cover anything you may appear lacking at
the time to solve your problem.

Most of us won't hold a grudge from one message to the next unless you
make similar mistakes after being warned. Even then, you may not have
any residuals from the last message. So even if you get flamed once or
twice, it doesn't diminish you in this community. I'm sure your parents
yelled at you many times as you grew up, yet you knew they loved you. We
usually will still accept you and help you even if we argue with you,
and even if we flame you.
 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] keeping Wiki in sync (new subject)

[Olle E. Johansson]

Philipp von Klitzing wrote:
I just added this to the SetVar() documentation on the Wiki. A question: 
If we keep on enhancing the Wiki step-by-step and day-by-day (I am 
silently in that game), how do we ensure that 

a) the information flows back into the documentation files of the CVS
That's something we have to discuss on the -doc list. I've started
adding some changes back to CVS for the Show application command.
We can't ensure that it's added. Also, the wiki allows for more text.

b) the Wiki information doesn't become outdated by newly introduced 
features and bug fixes?
We need to start documenting which version we document on the wiki.

Right now I try to stick to the CVS versions if nothing else is said
on the page. When the CVS split up, we need to document what applies
to stable and what applies to dev-branch.
So, bug fixes are not documented. Sometimes, we point to a bug report,
but it shouldn't be documented as it is part of the software.
We all have to watch the CVS update flow to ensure that we are synched,
this also applies to the doc project. The Wiki is community documentation.
As of know, there's no formal process or editor. It just seems to work.
Philipp, thank you for your contributions to the Wiki!

/O

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Re: [Asterisk-Users] Video

[Siggi Langauf]

On Thu, 1 Jan 2004, Olle E. Johansson wrote:

> [EMAIL PROTECTED] wrote:
> > What is the best software video client that is compatible with * ?
> To add a follow-up?
> Which channels support video and how?
> I know that there's support in H.323 and SIP. Anything else?

IAX (and IAX2), of course.
gnophone supports this, for example. Though gnophone isn't exactly
well-maintained, these days...

> There's nothing documented here on the Wiki, a black hole that needs to disappear ;-)
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Re: [Asterisk-Users] Programming an unlocked ADSI phone?

[Darren Nickerson]

Ken,

I'm able to reproduce that, unfortunately. Not all soft keys seem to cause
it to go dead, but some do. The phone doesn't exactly go dead, but the audio
does ... and the LCD begins to get pretty confused.

-d

--
Darren Nickerson
Senior Sales & Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 ext 8106 office
+1.215.243.8335 fax

- Original Message - 
From: "Ken Godee" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 01, 2004 2:51 PM
Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone?


> Darren Nickerson wrote:
> > That worked a treat - thanks! Comedian Mail is now able to download to
the
> > handset and there's a lot more functionality now.
> >
> > -d
>
> I'd be interested in knowing if once you try to use Comedian mail
> softkeys if the 480 keypad goes dead?
>
> Mine and several others reported same, which makes it useless, a shame
> to, I like the 480's ADSI function and haven't had a whole lot of time
> to look into it.
>
>
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Re: [Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank now Alcatel

[Lance Arbuckle]

TC wrote:
> 
> Hi
> I just came accross this
> Newbridge Mainstreet 3624 but the Alctel site  appears to have zip for
> reference/user manuals
> Anyone by chance have 1 of these or a url for the docs ?

Anyone know how to reset the passwords on the 3624 ?

Also, is the 3624 suitable for use with the T100P and Asterisk.  I was
considering one of these for my first asterisk channel bank since they
show up on Ebay regularly and fiarly cheap.

-- 
  .~.Triad Internet Systems, Inc.
  /V\Lance C. Arbuckle
 // \\   3315 Anderson Drive
/(   )\  Winston-Salem, NC 27127
 ^'~'^   336-771-2090
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Re: [Asterisk-Users] Programming an unlocked ADSI phone?

[Ken Godee]

Darren Nickerson wrote:
That worked a treat - thanks! Comedian Mail is now able to download to the
handset and there's a lot more functionality now.
-d
I'd be interested in knowing if once you try to use Comedian mail
softkeys if the 480 keypad goes dead?
Mine and several others reported same, which makes it useless, a shame 
to, I like the 480's ADSI function and haven't had a whole lot of time 
to look into it.

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Re: [Asterisk-Users] after hours - is this logic ok ?

[Lance Arbuckle]

Tilghman Lesher wrote:
> 
> On Thursday 01 January 2004 12:45, Lance Arbuckle wrote:

> > So, here's another question...  How does someone with no programming
> > experience, effectively learn the proper way to do things in Asterisk
> > ?
> 
> Trial and error is especially effective.  This is how I learned.  In
> addition, I read the source regularly, which is not nearly as daunting
> as it may seem, when taken in small doses.  Don't try to understand
> everything at once, but take a look at a small application, for example,
> SayUnixTime and trace back its function calls elsewhere into the code,
> into say.c and into stdtime/localtime.c, if you want to go that far.
> "grep -r some_function_name /usr/src/asterisk" will help you do the
> trace through code.  When somebody asks me about some functionality,
> if I don't know, the first thing I do is to go look in the source and
> see if I can figure it out.
> 
> For the variables, see the README.variables in the root directory of the
> Asterisk source.  The example I quoted above uses both variable
> interpolation ${} as well as expression evaluation $[], although the
> expression is the most simple.  You can also do comparisons in there:
> $[${var} > 3] or $[${var} = "oink"].
> 

well, I must admit I don't even attemp to read the source as I thought
it would be a waste of time since I don't know the language.  I see from
the header at the top of sayunixtime.c you know a little about that app
:)  Me thinks I need to go take an Intro to C class

-- 
  .~.
  /V\Lance C. Arbuckle
 // \\   
/(   )\  
 ^'~'^
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[Asterisk-Users] Newbridge Mainstreet 3624 T1 channel bank now Alcatel

[TC]

Hi
I just came accross this
Newbridge Mainstreet 3624 but the Alctel site  appears to have zip for
reference/user manuals
Anyone by chance have 1 of these or a url for the docs ?

thx

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RE: [Asterisk-Users] New to asterisk? RUN... don't walk.

[Brian West]

You said it good Look what this person posted to my blog... Now thats
what I call grown up.

Date: Thu, 1 Jan 2004 10:10:24 -0600
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]

IP Address: 24.10.200.168
Name: Jeff Sowery
Email Address: [EMAIL PROTECTED]
URL:

Comments:

You're a complete idiot.  Grow a brain or at least some balls.

-Jeff


NEXT!!!

bkw


On Thu, 1 Jan 2004, JR Richardson wrote:

> Piping in 2 cents,
>
> This is a great example of the Internet, Fast Food generation, showing their
> appreciation for all the magic that happens in the labs, hearts and minds of
> the courageous, hard working, dedicated and motivated group of people truly
> interested and guided to accomplish greatness.
>
> This platform for learning is one of the best tools in existence to come to
> a finite understanding of VoIP and legacy telephony with the versatility to
> expand beyond and develop originality in the field of telecommunications
> excellence, product development.  Learn it, understand it, appreciate it,
> then take it past where you found it and if you're capable contribute, if
> not, enjoy it.  But always, always maintain respect for those who created it
> and continue to refine it.
>
> Learning is intrinsically human, and in this world of Industry ("There is no
> substitution for knowledge." [Edward Deming]).  Find your inner child,
> re-capture and embrace what God has given you, the ability to learn.  It
> will require you to put down the remote control, get off the couch and
> decrease your apparently frequent visits to McDonalds.  Search and find the
> knowledge which you seek to ultimately fulfill your destiny; build an
> Asterisk Server that works.
>
> Hell, we all did.
>
> JR
>
>
>
>
>
>
> > Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST)
> > From: Me <[EMAIL PROTECTED]>
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] New to asterisk?  RUN... don't walk.
> > Reply-To: [EMAIL PROTECTED]
> >
> > As a newcomer to Asterisk, you will not be welcomed
> > with open arms.  First, you will find almost no
> > documentation on it's features.  Second, if you try to
> > ask questions, you will be flamed and pointed to
> > worthless how-tos and 'the wiki'.  These worthless
> > documents can only be useful for explaining how things
> > work to those already in-the-know.  Lastly, Asterisk
> > is so bug ridden, expect frequent segmentation faults.
> >  With a community so 'anti-n00b', don't expect your
> > problems to be fixed anytime soon.
> >
> > RUN!!! Don't walk... away from Aterisk.
> >
>
>
>
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>
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Re: [Asterisk-Users] after hours - is this logic ok ?

[Tilghman Lesher]

On Thursday 01 January 2004 12:45, Lance Arbuckle wrote:
> Tilghman Lesher wrote:
> > I'm curious as to why you're going the extra step to put a value
> > into the database.  Why not just set a channel variable and check
> > that value?
> >
> > [day]
> > exten => s,2,SetVar(daytime=1)
> >
> > [night]
> > exten => s,2,SetVar(daytime=0)
> >
> > [macro-stdexten]
> > exten => s,10,GotoIf($[${daytime}]?11:111)
> >
> > Using a variable is nice, because all channels get their own
> > variable space, so there's no chance of collisions between
> > different calls, and the variable is automatically trashed when the
> > channel is hungup.
> >
> > -Tilghman
>
> Hi Tilghman
> thanks for your response.
> I stuck the value in the database because I haven't a clue as to what
> I'm doing :-)  In all my reading I didn't see anything that talked
> about variables having their own private per channel sandbox in which
> to play.  I just figured since * had a DB, I'd use it :)  It's
> exactly this type of thing that I'm finding frustrating while trying
> to learn Asterisk.  I'll spend hours reading about something ( in
> this case I read a bunch about setvar and the DB commands ) but at
> the end of all that reading all I could show for it was two methods
> of accomplishing a task, but I din't know which way (if any) was the
> best or most appropriate way.  I feel like I'm handicapped in
> learning Asterisk by my lack of not knowing a programming language. 
> I have a sneaky feeling that a lot of the answers to my questions
> would be obvious to someone with C/C++ experience.
>
> So, here's another question...  How does someone with no programming
> experience, effectively learn the proper way to do things in Asterisk
> ?

Trial and error is especially effective.  This is how I learned.  In
addition, I read the source regularly, which is not nearly as daunting
as it may seem, when taken in small doses.  Don't try to understand
everything at once, but take a look at a small application, for example,
SayUnixTime and trace back its function calls elsewhere into the code,
into say.c and into stdtime/localtime.c, if you want to go that far.
"grep -r some_function_name /usr/src/asterisk" will help you do the
trace through code.  When somebody asks me about some functionality,
if I don't know, the first thing I do is to go look in the source and
see if I can figure it out.

For the variables, see the README.variables in the root directory of the
Asterisk source.  The example I quoted above uses both variable
interpolation ${} as well as expression evaluation $[], although the
expression is the most simple.  You can also do comparisons in there:
$[${var} > 3] or $[${var} = "oink"].

-Tilghman

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Re: [Asterisk-Users] after hours - is this logic ok ?

[Philipp von Klitzing]

Hi!

> > Using a variable is nice, because all channels get their own variable
> > space, so there's no chance of collisions between different calls,
> and > the variable is automatically trashed when the channel is hungup.

I just added this to the SetVar() documentation on the Wiki. A question: 
If we keep on enhancing the Wiki step-by-step and day-by-day (I am 
silently in that game), how do we ensure that 

a) the information flows back into the documentation files of the CVS
b) the Wiki information doesn't become outdated by newly introduced 
features and bug fixes?

> I stuck the value in the database because I haven't a clue as to what
> I'm doing :-)  In all my reading I didn't see anything that talked about
> variables having their own private per channel sandbox in which to
> play.  

You can find this in README.variables which is also included in the Wiki. 
However I agree that this fact is well hidden and hard to find or 
understand for a non-programmer. It's good that you pointed this out, and 
maybe next time you can help the community yourself by adding a comment 
to the Wiki or editing the relevant page. Sometimes newbies see thing 
that others have become blind for...

> I just figured since * had a DB, I'd use it :)  It's exactly this type
> of thing that I'm finding frustrating while trying to learn Asterisk. 
> I'll spend hours reading about something ( in this case I read a bunch
> about setvar and the DB commands ) but at the end of all that reading
> all I could show for it was two methods of accomplishing a task, but I
> din't know which way (if any) was the best or most appropriate way.  I
> feel like I'm handicapped in learning Asterisk by my lack of not
> knowing a programming language.  I have a sneaky feeling that a lot of
> the answers to my questions would be obvious to someone with C/C++
> experience. 

Learning about extensions.conf is indeed a little bit like learning a 
programming language. Anyway, as long as you have no need for more 
advanced functionality that requires AGI or other scripts I think you'll 
still be able to remain happy with what * offers.

Apart from that: As you know the handbook draft gives you a good start, 
and a more comprehensive book is under way that goes beyond a dry 
reference guide type document like the Wiki.

Cheers, Philipp


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Re: [Asterisk-Users] Programming an unlocked ADSI phone?

[Darren Nickerson]

That worked a treat - thanks! Comedian Mail is now able to download to the
handset and there's a lot more functionality now.

-d

--
Darren Nickerson
Senior Sales & Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 ext 8106 office
+1.215.243.8335 fax

- Original Message - 
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 01, 2004 12:38 PM
Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone?


> On Wednesday 31 December 2003 17:51, Darren Nickerson wrote:
> > Thanks 'gcc', that's exactly where I'm at now, with the exception of
> > the helpful comments on how to clear the services - thanks for those.
> > I already have those codes, and have used them to download (the
> > somewhat disappointing) 'asterisk.adsi' sample into both slots.
> >
> > The problem now is that Comedian Mail is trying to do fancy ADSI
> > stuff (because it can now), and a good deal of it fails because as
> > opposed to the app_adsiprog module which grabs lock code and FDN from
> > the adsi script itself, app_voicemail doesn't seem to allow me to set
> > those codes. It tries to do an FDM download at the beginning but
> > fails, since it seems to have a harcoded security code and fdn as
> > follows:
> >
> > static char *adapp = "CoMa";
> >
> > static char *adsec = "_AST";
> >
> > I tried changing these to
> >
> > static char *adapp = "0x85EFD9DA";
> >
> > static char *adsec = "0x78921D49";
> >
> > (the values suggested for the second slot) and recompiled
> > app_voicemail, but this still does not seem to be able to download to
> > the phone when I dial voicemail.
> >
> > I'm not a skilled C programmer (those folks are sane enough to have
> > left work to celebrate already), so there may be other magic I'm
> > missing ... but in summary it seems like app_voicemail doesn't have
> > any intelligence built into it to allow it to unlock phones as
> > required, even if the lock codes are available.
>
> static char *adapp = "\x85\xEF\xD9\xDA";
> static char *adsec = "\x78\x92\x1D\x49";
>
> -Tilghman
>
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>

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[Asterisk-Users] asterisk gateway to other gateways

[SW]

I was wondering why bother with this dialer thing at all :)

When you implement a new thing, it is a good idea to go back to basics and
think what you want to do.

You have multiple voip carriers and you have bunch of users. Based on the
number dialed you want to pick a carrier and dial out. May be authenticate a
certain caller for certain path. Then end of the day look for CDR's.

You do not need to set your call setup this fragmented.

How about this;

(a) User dials a international number
(b) * looks at the CallerID and Destination Number (at one database lookup)
(c) This caller is not allowed to dial this destination, so play a message
for that extent
(d) If caller is allowed, then * finds the least cost root and dial through
that path and connect the call.

* can do all these. So I would first sit and right down what I want my
system to do (in my users perspective). Then try to find a way to implement.
I wouldn't try to replace a box with another box.

my two cents 

SW


Message: 4
From: lito lampitoc <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Date: Fri, 02 Jan 2004 01:04:40 +0800
Subject: [Asterisk-Users] asterisk gateway to other gateways
Reply-To: [EMAIL PROTECTED]


Though I've had implementations of Asterisk, I havent encountered this
one yet, so i'd like to seek your advise if this possible.

I would want asterisk to be stand between the dialer the  destination.
The dialer will now dial asterisk access number. Asterisk will
acknowledge user by using CallerID and check against its database for
authentication and then sends out a DTMF A tone for ½ second to enable
the dialer to send the whole overseas digit.

Assume the caller is not in database, asterik could give user a busy
tone, IVR or just leave it and sends out a DTMF A tone anyway.


Once the overseas digit are sent from dialer to asterik, asterik will
then decide which telco/carrier/Voip to send the traffic to using LCR.
Please note that we need to assign at least 5-10 telco/carrier/Voip
access number for backup purposes.


Once the least cost destination is selected by asterik, asterik will
pick up the PRI line and dial a local access number and waits for a DTMF
A tone. Once the A tone is heard from telco/carrier/Voip, it will send
the overseas digit which was sent by the dialer earlier on.


Also, can asterik sends out a musical tone or IVR while connecting to
other telco to advice user that the call is connecting, else it would be
dead air from there on.


The whole process takes less than 5 seconds while the user stays on the
line for this whole thing to happen.


Thanks.


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Re: [Asterisk-Users] after hours - is this logic ok ?

[Lance Arbuckle]

Tilghman Lesher wrote:

> 
> I'm curious as to why you're going the extra step to put a value into
> the database.  Why not just set a channel variable and check that value?
> 
> [day]
> exten => s,2,SetVar(daytime=1)
> 
> [night]
> exten => s,2,SetVar(daytime=0)
> 
> [macro-stdexten]
> exten => s,10,GotoIf($[${daytime}]?11:111)
> 
> Using a variable is nice, because all channels get their own variable
> space, so there's no chance of collisions between different calls, and
> the variable is automatically trashed when the channel is hungup.
> 
> -Tilghman

Hi Tilghman
thanks for your response.
I stuck the value in the database because I haven't a clue as to what
I'm doing :-)  In all my reading I didn't see anything that talked about
variables having their own private per channel sandbox in which to
play.  I just figured since * had a DB, I'd use it :)  It's exactly this
type of thing that I'm finding frustrating while trying to learn
Asterisk.  I'll spend hours reading about something ( in this case I
read a bunch about setvar and the DB commands ) but at the end of all
that reading all I could show for it was two methods of accomplishing a
task, but I din't know which way (if any) was the best or most
appropriate way.  I feel like I'm handicapped in learning Asterisk by my
lack of not knowing a programming language.  I have a sneaky feeling
that a lot of the answers to my questions would be obvious to someone
with C/C++ experience.

So, here's another question...  How does someone with no programming
experience, effectively learn the proper way to do things in Asterisk ?

For example, if I want to implement feature xyz, currently I go googling
and read, read, read.  I may come up with two or three sample contexts
that accomplish the same thing via different means.  All I can do now is
pick one of the examples and modify to my needs without any regard to
the pros and cons of each example.  Now, I *try* to post intelligently
phrased questions to this list but after seeing some of the threads the
past few days, I'm hesitant to post the "how do I implement this?"
questions for fear of being flamed to death :( or getting the standard
"Google is your friend" responses.  What would be a great help to me and
possible other beginners, is whenever a question is asked of the list,
responses include a brief explanation of the "why" for the particular
answer.  Or, if there is another resource that talks in generalities
about how to do thing in Asterisk, please, I beg the list, to tell me
where it is :)


Happy New Year !!

-- 
  .~.
  /V\Lance "I'm always looking for clue's" Arbuckle
 // \\   
/(   )\  
 ^'~'^
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[Asterisk-Users] * crash when forward voicemail --Nicolas Gudino

[JR Richardson]

Hey Nicolas,

That did it.  I ran that export command you suggested, then launched *,
everything worked fine.  I'm still looking for info on what that command
actually does.  Can you shed some light please?

Thanks.

JR

-Original Message-
From: JR Richardson [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, December 30, 2003 6:44 PM
To: '[EMAIL PROTECTED]'
Subject: Re: * crash when forward voicemail message [problem solved]



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, December 30, 2003 6:21 PM
No I didn't, I don't have a clue what that is or does.  Please explain, I'll
try it and let you know.

Thanks.

JR

Did you try with this line before launching asterisk (with stock redhat
9 kernels):

export LD_ASSUME_KERNEL=2.4.1

Best regards,

On Tue, 2003-12-30 at 20:07, JR Richardson wrote:
> Thanks for all your help Martin,
> 
> Guys,
> 
> This is a good find and hopefully could help someone else.
> 
> I've been having a problem with forwarding voicemail from one mailbox to
> another.  I ran down the sendmail and soundcard path and came up goose
eggs.
> With intuitive guidance from Martin Pycko (Digium), I switched from Redhat
9
> Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it seemed to
> solve the problem I was having.  There is still a little weirdness going
on
> but the voicemail forward command is working.  During a -dgc session,
I




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Re: [Asterisk-Users] Programming an unlocked ADSI phone?

[Tilghman Lesher]

On Wednesday 31 December 2003 17:51, Darren Nickerson wrote:
> Thanks 'gcc', that's exactly where I'm at now, with the exception of
> the helpful comments on how to clear the services - thanks for those.
> I already have those codes, and have used them to download (the
> somewhat disappointing) 'asterisk.adsi' sample into both slots.
>
> The problem now is that Comedian Mail is trying to do fancy ADSI
> stuff (because it can now), and a good deal of it fails because as
> opposed to the app_adsiprog module which grabs lock code and FDN from
> the adsi script itself, app_voicemail doesn't seem to allow me to set
> those codes. It tries to do an FDM download at the beginning but
> fails, since it seems to have a harcoded security code and fdn as
> follows:
>
> static char *adapp = "CoMa";
>
> static char *adsec = "_AST";
>
> I tried changing these to
>
> static char *adapp = "0x85EFD9DA";
>
> static char *adsec = "0x78921D49";
>
> (the values suggested for the second slot) and recompiled
> app_voicemail, but this still does not seem to be able to download to
> the phone when I dial voicemail.
>
> I'm not a skilled C programmer (those folks are sane enough to have
> left work to celebrate already), so there may be other magic I'm
> missing ... but in summary it seems like app_voicemail doesn't have
> any intelligence built into it to allow it to unlock phones as
> required, even if the lock codes are available.

static char *adapp = "\x85\xEF\xD9\xDA";
static char *adsec = "\x78\x92\x1D\x49";

-Tilghman

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Re: [Asterisk-Users] after hours - is this logic ok ?

[Tilghman Lesher]

On Wednesday 31 December 2003 19:16, Lance Arbuckle wrote:
> Andrew Thompson wrote:
> > - Original Message -
> > From: "Lance Arbuckle" <[EMAIL PROTECTED]>
> > >
> > > Before I get too excited, I wanted to get all you experts to look
> > > at the how I implemented my after hours test.  The goal is to
> > > prevent the phone from ringing afer certain hours, just go to VM.



> [day]
> exten => s,2,DBput(FEATURE/DAY=yes)
> exten => s,3,Goto(s,10)
>
> [night]
> exten => s,2,DBdel(FEATURE/DAY) ;if we got here it must be night
> time so remove the key
> exten => s,3,Goto(s,10)
>
> [macro-stdexten]
> exten => s,1,NoOp
> < other testing crap deleted >
> exten => s,10,DBget(foo=FEATURE/DAY); is it day time ?  if key
> exists, goto n+1, otherwise n+101
> exten => s,11,Goto(s,201) ; yes, well let's ring the phones
> exten => s,111,Goto(s,204)  ; no, goto uVM

I'm curious as to why you're going the extra step to put a value into
the database.  Why not just set a channel variable and check that value?

[day]
exten => s,2,SetVar(daytime=1)

[night]
exten => s,2,SetVar(daytime=0)

[macro-stdexten]
exten => s,10,GotoIf($[${daytime}]?11:111)

Using a variable is nice, because all channels get their own variable
space, so there's no chance of collisions between different calls, and
the variable is automatically trashed when the channel is hungup.

-Tilghman

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[Asterisk-Users] help

[Shanon Swafford]

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[Asterisk-Users] asterisk gateway to other gateways

[lito lampitoc]

 
Though I've had implementations of Asterisk, I havent encountered this
one yet, so i'd like to seek your advise if this possible.

I would want asterisk to be stand between the dialer the  destination.
The dialer will now dial asterisk access number. Asterisk will
acknowledge user by using CallerID and check against its database for
authentication and then sends out a DTMF A tone for  second to enable
the dialer to send the whole overseas digit.

Assume the caller is not in database, asterik could give user a busy
tone, IVR or just leave it and sends out a DTMF A tone anyway.

 
Once the overseas digit are sent from dialer to asterik, asterik will
then decide which telco/carrier/Voip to send the traffic to using LCR.
Please note that we need to assign at least 5-10 telco/carrier/Voip
access number for backup purposes.

 
Once the least cost destination is selected by asterik, asterik will
pick up the PRI line and dial a local access number and waits for a DTMF
A tone. Once the A tone is heard from telco/carrier/Voip, it will send
the overseas digit which was sent by the dialer earlier on.

 
Also, can asterik sends out a musical tone or IVR while connecting to
other telco to advice user that the call is connecting, else it would be
dead air from there on.

 
The whole process takes less than 5 seconds while the user stays on the
line for this whole thing to happen.
 

Thanks.

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Re: [Asterisk-Users] Java?

[Philipp von Klitzing]

Hi!

> > We needed the client browser to be open all the time for dynamic data to
> > load without the page refreshing. After looking at all of our options we
> > decided on programming it ourselves using flash rather than java. 
> 
> Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good.
> Dynamic effective,Easy coding and Fast response :-)

That's an excellent suggestion, I agree with Ray. Masakazu, do you think 
you could provide a working sample either here on the list or in the 
Wiki?

Cheers, Philipp


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Re: [Asterisk-Users] New to asterisk? RUN... don't walk.

[WipeOut]

JR Richardson wrote:

Piping in 2 cents,

This is a great example of the Internet, Fast Food generation, showing their
appreciation for all the magic that happens in the labs, hearts and minds of
the courageous, hard working, dedicated and motivated group of people truly
interested and guided to accomplish greatness.
This platform for learning is one of the best tools in existence to come to
a finite understanding of VoIP and legacy telephony with the versatility to
expand beyond and develop originality in the field of telecommunications
excellence, product development.  Learn it, understand it, appreciate it,
then take it past where you found it and if you're capable contribute, if
not, enjoy it.  But always, always maintain respect for those who created it
and continue to refine it.
Learning is intrinsically human, and in this world of Industry ("There is no
substitution for knowledge." [Edward Deming]).  Find your inner child,
re-capture and embrace what God has given you, the ability to learn.  It
will require you to put down the remote control, get off the couch and
decrease your apparently frequent visits to McDonalds.  Search and find the
knowledge which you seek to ultimately fulfill your destiny; build an
Asterisk Server that works.
Hell, we all did.

JR

 

BRAVO!!! Well said!!

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RE: [Asterisk-Users] Java? --> Ming!

[Ray Burkholder]

> Masakazu Nakano
> Sent: December 31, 2003 21:13
> 
> On Wed, 31 Dec 2003 21:19:10 +0200
> "Stephen Karrington" <[EMAIL PROTECTED]> wrote:
> 
> > We needed the client browser to be open all the time for 
> dynamic data to
> > load without the page refreshing. After looking at all of 
> our options we
> > decided on programming it ourselves using flash rather than java. 
> snip
> 
> Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good.
> 
> Dynamic effective,Easy coding and Fast response :-)
> 

Cool.  I like the Ming thing.  Also works with Perl (many Perl examples
available).  And has an XML event interface for two way communications with
a server.  Certainly is way much less overhead than the Java thing I was
contemplating.

Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 644 6999 x2002

P.S.  Note, for the message police, I cut out extraneous text, did the
attribution at the top, did a bottom post, and made it a single page for
zero scrolling.


-- 
Scanned for viruses and dangerous content at 
http://www.oneunified.net and is believed to be clean.

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Re: [Asterisk-Users] asterisk reload for FWD to register

[Philipp von Klitzing]

Hi!

> so asterisk runs fine but it doesnt show attempts or try nothing
> similar to register with FWD, i have to manually log in the console and
> reload asterisk by "reload", after that it shows its trys to register
> with FWD and finally makes it and recives incoming calls from FWD. 

Two things:

- do a "sip debug" to check what exactly is happening. Often there is SIP 
activity going on that doesn't show in the CLI.

- take a look at bug 657 and add a bug note if you think this is related:
http://bugs.digium.com/bug_view_page.php?bug_id=657

Cheers, Philipp


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RE: [Asterisk-Users] New to asterisk? RUN... don't walk.

[JR Richardson]

Piping in 2 cents,

This is a great example of the Internet, Fast Food generation, showing their
appreciation for all the magic that happens in the labs, hearts and minds of
the courageous, hard working, dedicated and motivated group of people truly
interested and guided to accomplish greatness.

This platform for learning is one of the best tools in existence to come to
a finite understanding of VoIP and legacy telephony with the versatility to
expand beyond and develop originality in the field of telecommunications
excellence, product development.  Learn it, understand it, appreciate it,
then take it past where you found it and if you're capable contribute, if
not, enjoy it.  But always, always maintain respect for those who created it
and continue to refine it.

Learning is intrinsically human, and in this world of Industry ("There is no
substitution for knowledge." [Edward Deming]).  Find your inner child,
re-capture and embrace what God has given you, the ability to learn.  It
will require you to put down the remote control, get off the couch and
decrease your apparently frequent visits to McDonalds.  Search and find the
knowledge which you seek to ultimately fulfill your destiny; build an
Asterisk Server that works.

Hell, we all did.

JR






> Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST)
> From: Me <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] New to asterisk?  RUN... don't walk.
> Reply-To: [EMAIL PROTECTED]
> 
> As a newcomer to Asterisk, you will not be welcomed
> with open arms.  First, you will find almost no
> documentation on it's features.  Second, if you try to
> ask questions, you will be flamed and pointed to
> worthless how-tos and 'the wiki'.  These worthless
> documents can only be useful for explaining how things
> work to those already in-the-know.  Lastly, Asterisk
> is so bug ridden, expect frequent segmentation faults.
>  With a community so 'anti-n00b', don't expect your
> problems to be fixed anytime soon.
> 
> RUN!!! Don't walk... away from Aterisk.
> 



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[Asterisk-Users] asterisk reload for FWD to register

[Miguel Cavazos]

hi,

guys I have a weird, really weird problem with FWD (free world dialup),
My serveris a P2 400mHz 64 on Ram. This server is setup only to answer
to its FWD # for a friend to do calls to its local PBX. When i boot up
the server running debian Woody, with kernel 2.4.23, running asterisk
CVS from DEC 30 2003, asterisk is loaded with no problems on the init.d,
so asterisk runs fine but it doesnt show attempts or try nothing similar
to register with FWD, i have to manually log in the console and reload
asterisk by "reload", after that it shows its trys to register with FWD
and finally makes it and recives incoming calls from FWD.

The problem here is I don't want to reload the server every time the box
goes for reboot so I wonder if someone is having this same problem or
had it before.

Miguel 
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Re: [Asterisk-Users] Video

[Olle E. Johansson]

[EMAIL PROTECTED] wrote:
What is the best software video client that is compatible with * ?
To add a follow-up?
Which channels support video and how?
I know that there's support in H.323 and SIP. Anything else?
There's nothing documented here on the Wiki, a black hole that needs to disappear ;-)

/O

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Re: AW: [Asterisk-Users] Snom 200 with two extns defined anyone?

[Olle E. Johansson]

Christian Stredicke wrote:
We at snom have problems with Asterisk when we receive calls without the
line indication. When we register we place a contact like this:
REGISTER sip:asterisk SIP/2.0
Contact: 
When we receive the 200 Ok, we search for the "h35h345". If we don’t find
it, we try to guess which line is affected. This is relatively easy on a
REGISTER response, but on an incoming INVITE we have serious problems. I
think some of the challenging and line-assignment problems are related to
this problem.
Strictly speaking, we register the contact
"sip:[EMAIL PROTECTED];line=h35h345", NOT "sip:[EMAIL PROTECTED]"! Parameters
are an essential part of a URI which must not be discarded.
Ok, Christian, let's fix this.

First, I'm curious, is the "line=" parameter specified somewhere? (Always looking
for documentation :-)
Secondly, in many places in the sip channel, everything after the ";" is discarded.
I would really appreciate if Snom could help us fixing this, so Snom phones work 
correctly
and fully with Asterisk. (Have a new Snom 200 on my desk :-)
I'm not an experienced C programmer, so I can't fix this myself. However, there are
experienced C programmers in the community that will fix this, but they need proper
and detailed input on what to fix.
I've opened a bug
http://bugs.digium.com/bug_view_page.php?bug_id=732
Let's continue adding information there.

BTW, there's some other Snom problems where we need input from SNOM. Search on "Snom"
in the bug tracker. Thank you for participating in the Asterisk community!
/O

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Re: [Asterisk-Users] New to asterisk? RUN... don't walk.

[Olle E. Johansson]

Andrew Kohlsmith wrote:

4. this mailing list's ARCHIVES. 
http://lists.digium.com/pipermail/asterisk-users/  you can search the 
archives by using google and including "site:lists.digium.com" in your 
search.

Or use http://search.voip-forum.com

/O :-)

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[Asterisk-Users] Video

[jna]

What is the best software video client that is compatible with * ?

John
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Re: [Asterisk-Users] Anyone, ideas for incoming call management for CRM system

[Olle E. Johansson]

Philipp von Klitzing wrote:
Hi!


Now I am looking at the other way around. If a call comes in, I want our 
web based system to automatically detect the number and present the call 
information to the user.

What I am looking for is a solution like this:
* Call comes in
* XXX on Line YYY answers
* A URL to a web page is transmitten on some channel, preferably the VoIP channel
* The web page opens in a web window´
I think it has been mentioned on the list that GnoPhone supports this, but I
can't find it now.
Anyone knows more on the Gnophone URL stuff mentioned here:
http://lists.digium.com/pipermail/asterisk-users/2003-November/026371.html
You could use this application:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SendURL
Anyone knows which channels support sending URLs?
SIP does, does this work with the Cisco 7960 phone?
Does IAX2 support this? If so, will DIAX support it? (hint, hint)

As said in previous mail, we could add more information to the wiki on these kind
of solutions. (And break the fellowship of the Ring :-)
/O

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RE: [Asterisk-Users] Re: Grandstream Early Dial

[Dave Cotton]

On Thu, 2004-01-01 at 03:17, Josh Roberson wrote:
> I've never had early dial working, however, I resolved my multiple digit
> issue by simply putting both the GS phones and asterisk in INFO mode.
> This worked on both 10.0.3.81 firmware on the budgetone and the ATA286,
> as well as 10.0.4.30 firmware.  I'm not saying I don't believe you, but
> doubelcheck your lines in asterisk to be dtmfmode=info and the gs
> devices are on SIP INFO method, and your DTMF Payload type is 101.

The really strange thing is that my voicemail system had worked
flawlessly before, with the 1.0.4.30 firmware. What had changed is that
I was running SER and had * and the GS on 5061 to avoid clashing with
SER on the same machine. Because I couldn't get SER compiled I
reconfigured back to 5060 and then my problems started.
-- 
Dave Cotton
Directeur
Linux Autrement
193 rue Marcel Cerdan
84270 Vedene
04 90 23 30 81

IAX 17004902330 FWD 42651

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AW: [Asterisk-Users] Snom 200 with two extns defined anyone?

[Christian Stredicke]

We at snom have problems with Asterisk when we receive calls without the
line indication. When we register we place a contact like this:

REGISTER sip:asterisk SIP/2.0
Contact: 

When we receive the 200 Ok, we search for the "h35h345". If we don’t find
it, we try to guess which line is affected. This is relatively easy on a
REGISTER response, but on an incoming INVITE we have serious problems. I
think some of the challenging and line-assignment problems are related to
this problem.

Strictly speaking, we register the contact
"sip:[EMAIL PROTECTED];line=h35h345", NOT "sip:[EMAIL PROTECTED]"! Parameters
are an essential part of a URI which must not be discarded.

BTW the line-id has also the function of increasing the security
significantly. Just imaging a network script that sprays INVITE packets
through the network - the whole office would start ringing.

#!/bin/bash
# Give the colleagues a wake-up call!
i=1
while [ $i -lt 256 ]; do
  echo INVITE sip:192.168.0.$i SIP/2.0^M >junk.txt
  echo f: sip:[EMAIL PROTECTED] >>junk.txt
  echo t: sip:[EMAIL PROTECTED] >>junk.txt
  echo i: [EMAIL PROTECTED] >>junk.txt
  echo l: 0^M >>junk.txt
  echo CSeq: 1 INVITE^M >>junk.txt
  echo ^M
>>junk.txt
  cat junk.txt >/dev/udp/192.168.0.$i/5060
  i=$[$i+1]
done

Unfortunately, the snom phones are not able to pick the right line on these
packets!

By requiring the line id, the phone would reject the INVITE as it can not
match the request URI with any valid line on the phone! BTW that’s why I
hate "direct IP calls" - it gives you zero security against such attacks.

Our latest image is 2.03e (http://snom.com/download/share), it does strict
line id comparison and tries to deal with "junk" line identifications as
good as possible. 


Happy New Year! Christian

> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] Im Auftrag von Rich Adamson
> Gesendet: Donnerstag, 1. Januar 2004 00:26
> An: Asterisk-a-users-list
> Betreff: [Asterisk-Users] Snom 200 with two extns defined anyone?
> 
> 
> Are there any Snom 200 users that have two extns defined on their phone?
> 
> I've been trying to get two (or more) extns defined in such a way that
> when extn #1 rings, LED #1 flashs; extn #2 rings, LED #2 flashes, etc.
> (Answer greating will be different depending upon which extension is
> called.)
> 
> I can get multiple extns to register and work with * just fine, but
> regardless of which extn is called, only the Line 1 key flashes. I'm
> tried both v2.02t and v2.03e code with same result.
> 
> As soon as I define Key Mappings for P1 and P2, the phone goes into
> a constant Register, 100 Trying, 407 Proxy Authentication Required loop
> that never ends. 1,000's of packets spewing in the loop.
> 
> Thoughts?
> 
> Rich
> 
> 
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Re: [Asterisk-Users] New to asterisk? RUN... don't walk.

[Balaji NJL]

i second it.

i had no previous experience with any telecom
equipment or even the lingo. I
didnt know what channel meant in telecom world. i
started with the *
handbook and did a lot of googling, searched the
archives. Search, search
and search. There is lot of info here. i am happy to
say that i hv my * up
and running.

* is a great product and Hats off to the team. Keep up
the good work.
-B

- Original Message - 
From: "Josh Roberson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, December 31, 2003 6:17 PM
Subject: RE: [Asterisk-Users] New to asterisk? RUN...
don't walk.


> Well, since everyone else is top-quoting on this
message, so will I :P
>
> I'm no veteran either.  As a matter of fact, I have
had ZERO prior
> knowledge to the telcom industry or more than 'user
level' experience
> with telecommunications in general.  I decided that
I wanted to expand
> my knowledge, and actually LEARN a few things, so I
jumped into
> asterisk.  I was, and quite frankly, IMO, still AM a
'n00b' to *.
> However, after playing around, and learning what
things do, by reading
> the documentation that IS there, searching the
archives, and just
> trolling the list and IRC, I have learned more in
the last 4-5 months of
> having * than a lot of people I've noticed have
learned in a lifetime of
> experience.I now have a fully functional (well,
minus MOH, because
> mpg123 isn't yet compiled on my new box), *
implementation, serving
> myself and my roommates strictly over VoIP, and a
couple ata's and a
> Internet PhoneJack card.  I love it.  And I'm STILL
learning to this
> date.
>
> Asterisk is not something you can expect everyone to
just drop what
> their doing and help you with.  Sure, it can be
frustrating, but if you
> are so dense that you can't sit down an play with it
and learn what
> happens when you type something in the cli, or
change a few things in
> your dialplan, then get out, I agree.
>
> If you liked taking apart mom's hairdryer as a kid
and seeing how it
> worked, and then later on, rewired up a few things
to do what you wanted
> them to, or even took a hex editor to command.com in
msdos to change
> what it says to suit your taste (mucho guilty on
that one.. lol), then
> you will have no problem finding out what you can
and can't change
> simply by editing files, and trying things out.
>
> Take off your training wheels, and just TRY IT.
>
> - Josh R.
> [EMAIL PROTECTED]
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of SW
> > Sent: Wednesday, December 31, 2003 4:13 PM
> > To: [EMAIL PROTECTED] Digium. Com
> > Subject: [Asterisk-Users] New to asterisk? RUN...
don't walk.
> >
> > Hello,
> >
> > I am not a veteran here, but would like to share
my thoughts on this
> > subject.
> >
> > True, * is opensource and freely available, but it
is not a computer
> > program
> > that you download and run. It is a very versatile
telecommunication
> > product
> > you would otherwise pay at least 100 K to buy from
a telecom vendor,
> if
> > not
> > more based on modules and usage, license
hash-codes etc.
> >
> > Even to try * one would need some pre requisite
knowledge in telecom,
> if
> > not
> > many years in the field. I work for a large
telecom company and my
> > specialty
> > is voice over broadband (or xDSL). I worked with
asterisk for couple
> of
> > months now and I am amazed to see areas of telecom
that * touch upon
> with.
> > Starting from Linux, to SIP, H323, DSL
technologies (PPP, PPPoE,
> PPPoA,
> > DHCP, NAT), Call routing(Dial Plan), IVR,
Transcoding, STUN are few
> areas
> > that one would have to master even thinking about
*.
> >
> > True one would know the syntax, and howtos etc,
but also would have to
> > have
> > the ability to troubleshoot. For last two-three
months in this list, I
> > have
> > not seen any newbi posting a sip trace (from a
ethereal or a TCP dump)
> and
> > asking a question about it. I have seen many
question for instance,
> asking
> > syntax of h.323 dial, but never seen a question
asked on a h323 trace.
> >
> > I think, having * openly available is like keeping
an airplane openly
> > available in a airfield, so that anybody can try
flying. Tell me how
> many
> > of
> > us would go try and fly that airplane if we do not
know how to fly :)
> >
> > Point that I want to make here is simple, please
try to understand
> what *
> > is
> > all about. If you like it's features and would
like it to run in a
> > production environment try to get some
professional help. If you are
> > learning these technologies for fun then get
educated, use tools
> available
> > to troubleshoot. Hooking up couple of phones and
making a call is far
> from
> > knowing *.
> >
> > Asterisk is a great product (thanks Mark and many
others) and if you
> know
> > what you are doing, you can do wonders with it.
Don't put it down,
> because
> > you do not have the background to understand it or
work with it.
> >
> > Cheers
> >