Re: [Asterisk-Users] IBM x306

[Stefan de Konink]

On Sun, 25 Sep 2005, Marco Supino wrote:

> I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my
> problem is that the BIOS assigns the same IRQ to the SCSI controller,
> and the TDM400P, i have tried several options of making the bios change
> the IRQ, but it will always move them together, anyone with some info
> about my options ?

Linux usually don't care about Bios settings, you could try kernel cmdline
parameters. Acpi and IRQ are google terms for it.


Stefan

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Re: [Asterisk-Users] SS7 support ?

[Stefan de Konink]

On Sat, 24 Sep 2005, Usman wrote:

> Is there any digium card that support E1 with SS7  and does Asterisk 
> support SS7 ???
>
> any 1 who has done this ?
Maybe google has?

http://www.google.nl/search?q=Asterisk+SS7&start=0&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-US:unofficial

He does: http://www.voip-info.org/tiki-index.php?page=Asterisk+SS7

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Re: [Asterisk-Users] internet connection between Africa and Europe

[Stefan de Konink]

Check out google with: VSAT Africa, lots of companies provide IP links
overthere. If it is good enough for voip... I don't yet know.

Stefan

On Thu, 15 Sep 2005, Jean-Michel Hiver wrote:

> Stéphane LAVRI a écrit :
>
> >Hi
> >
> >I'm looking for a company who can provide me an Internet connection
> >between africa and Europe.
> >
> >
> 'Africa' and 'Europe' are both rather big, so what you're saying doesn't
> make much sense. Pehaps if you outlined your requirements a bit better,
> you could get some useful advice.
>
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Re: [Asterisk-Users] Streaming MP3's from Asterisk with Ices

[Stefan de Konink]

On Tue, 19 Jul 2005, Geoff Karl wrote:

> >From what i can see in the ices configuration there is no way to get
> an input other than an mp3 playlist.  In order to work with Asterisk I
> need to use the stdinpcm input module.
>
> I am sure someone has a solution to get mp3 audio out of asterisk.

What about ezstream, you can pipe audio in and stream it to icecast2?


Stefan

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Re: [Asterisk-Users] OT: Congrats, Europe!

[Stefan de Konink]

On Thu, 7 Jul 2005, Anton Tinchev wrote:

> Vahan Yerkanian wrote:
>
> > http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150&tid=147&tid=136
> >
> > http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
> So we're are waiting the free g729 codec for Europe now ...

No need for celebration... http://www.cnn.com/

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Re: [Asterisk-Users] IAX aproprietary protocol

[Stefan de Konink]

On Wed, 27 Apr 2005, Joseph wrote:
> How can proprietary protocol be open protocol?

If the protocol is fully documentated and this documententation is
available to anyone you can speak of a open protocol. It is not an open
'standard', because it is only supported by Digium, thus proprietary.

http://en.wikipedia.org/wiki/Proprietary

Stefan

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Re: [Asterisk-Users] Small PHP script for displaying * CID database in Cisco 7940/60 XML

[Stefan de Konink]

On Mon, 7 Feb 2005, Stefan Gofferje wrote:

> I have written a small PHP-script to display the * CID database as a
> telephone directory in Cisco 7940/7960 XML-displays.
>
> Find it at http://gofferje.homelinux.org/download/directory.phps

Did you get the .php extension working in the 7940/60 or does this still
require renaming the file to .cgi? (With handler support)


Greetings,

Stefan de Konink

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Re: [Asterisk-Users] Asterisk as root in realtime vs. non-rootasterisk?

[Stefan de Konink]

On Thu, 27 Jan 2005, Robert Rozman wrote:
> I mean running Asterisk as pseudo realtime thread with -p command line
> argument - this is only possible if Asterisk is run as root...

In order to drop privileges asterisk must be started as root too, doesn't
 it? So if Asterisk gets into pseudo realtime before it drops privileges
and switches to an other user/group there would be no problem at all.

I'm under the assumption that i'm running pseudo realtime with
asterisk/voip, must admit: never checked it.


Stefan de Konink

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Re: [Asterisk-Users] Call back when no longer busy

[Stefan de Konink]

E. Versaevel wrote:
Hello, I’m trying to implement a function available on the PSTN net 
here, if you dial a number which is busy and you press 5, you will be 
called back when the busy party hangs up.
Cron job, eighter parse every 10s both peer statuses, and create a call 
file that is in the callback context which parses the from peer callback 
database?

Or in a ideal situation an application that notices peer changes and 
creates a trigger. (Manager API Events)

Wait for Hangup, check Hangup->Channel in To5-DB
if in call Callback context
(extension script should handle pickup/call To5-DB/FromPeer)
Greetings,
Stefan de Konink
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Re: [Asterisk-Users] [Asterisk-Dev] RE: [Asterisk-biz] Asterisktraining andcertification :: AstriconTraining

[Stefan de Konink]

Hi,
dean collins wrote:
Here here, a bit of documentation online would cut down considerably the
traffic on this list asking how to questions.
There is enough information "online" at the moment probably the ability 
to find/search your anwers is very limited. First above anything Wiki's 
are good, it makes it easy for all kind of people to put something 
online. In an ideal situation you want a Wiki 'Expert System' before 
your mailinglist. There is much information online but nobody seems to 
know, or is able to find it. This is not Asterisk related but more a 
general problem of the 21th century.

Microsoft put a kind of question-answer system inside Windows98 (and 
maybe 95) maybe it is not that bad at all to cut down traffic.

I pick a random example:
Can anybody explain me the difference between monitoring and recording?
Can I record a conversation during a phone call, e.g., I find that I should note that, and press 
"7" to record till I press again "7" (or whatever key is good for that.) while 
the rest of the phone call will be not recorded!
bye
Ronald
$ What is your problem related to?
> difference between monitoring and recording
$ Answer: The actions monitoring and recording differ, while cmd monitor 
records incomming and outgoing channels to different soundfiles, cmd 
record acts as a prompt for voice input.
$ Is this the answer you are looking for?
> Yes, show examples

$ Good;
cmd monitor ->
 exten => 2060,1,Answer
 exten => 2060,2,Wait(1)
 exten => 2060,3,Monitor(wav,myfilename)
 exten => 2060,4,Meetme(1,ps)
cmd Record ->
[recordings]
exten => 500,1,Festival('Please record your message')
exten => 500,2,Record(mymessage:gsm)
exten => 500,3,Festival('You said')
exten => 500,4,Playback(mymessage)
exten => 500,5,Festival('Press 1 to continue or 2 to change your message')
exten => 500,6,ResponseTimeout(3)
exten => t,1,Festival('Sorry, I did not get that')
exten => t,2,Goto(500,5)
exten => i,1,Festival('Sorry, that is an invalid choice')
exten => i,2,Goto(500,5)
exten => 1,1,System(/bin/mv /var/lib/asterisk/sounds/mymessage.gsm 
/var/lib/asterisk/sounds/local/${TIMESTAMP}.gsm)
exten => 1,2,Festival('Thank you, your recording has been saved.')
exten => 1,3,Festival('Press 3 to record another file or 4 to hang up')

exten => 2,1,Goto(500,1)
> explain cmd Record example
$ Which lines?
.
(I have no clue if this answer is correct but it is the way I 
interpreted the wiki pages 
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Record and 
http://www.voip-info.org/wiki-Asterisk+cmd+monitor)

This isn't even futuristic and probably a xmpp://[EMAIL PROTECTED] 
makes a nice jabber interface to it.

Cmon sokal, put the questions and training material online.
I think there is effort in this question/training material and if it was 
released under the GNU Free Documentation License many people would be 
happy. Other people would be able to contribute cases/questions/etc, but 
from a business point of view somebody started this on their own (which 
looks to me a bit strange in a OSS-community), and now have a 'real' 
Asterisk certification for it, so don't bother if you don't like it, 
ofcourse you/we can start a Free Online Asterisk Certification Program.

Someone once told me something about cisco certs, cisco prefers to hire 
non-cisco certified people over cisco-certified people because they 
learned by theirselves the knowledge required to work there. (He was 
core designer... without any certs). Don't know if it is true, but I do 
like the spirit behind it.


Stefan de Konink
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Re: [Asterisk-Users] Dynamically Choose Codec for Bandwidth Management

[Stefan de Konink]

[EMAIL PROTECTED] wrote:
Is there any way to set Asterisk to choose what codec to allow for a new 
call based on current usage?
I think there is a way. Since I'm not in the stage yet to configure my 
extensions.conf on that deep level I found some clues.

http://www.voip-info.org/wiki-Asterisk+variables
${SIP_CODEC}: Used to set the SIP codec for a call
Probably if you make the call go thru an extension which checks current 
bandwidth consumption via an external program. (Something AGI) You could 
make the call jump to an low/normal/high bandwidth setting by set the 
SIP_CODEC for the to be used codec. With a bit of magic you probably can 
 check the amount of free G729 licences too.

Greetings,
Stefan de Konink
ps. The idea is neat... I'm definately going to try to work out some code.
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Re: [Asterisk-Users] Website that reads text recently on the list?

[Stefan de Konink]

Steve Totaro wrote:
there was a website on the list recently that allowed you to enter text 
(up to 50 words) and it would create a wav file with various voice 
options.  does anyone remember what it was?  rapsody something or another.
http://www.babeltech.com/Demos.php?s=48&m=3&f=95
http://www.scansoft.com/realspeak/demo/
Stefan de Konink
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Re: [Asterisk-Users] High(er) availability

[Stefan de Konink]

E. Versaevel wrote:
Which app do you use for monitoring the primary box and if it fails
taking over the IP address by the backup one? I haven't found a suitable
(active-active) app so far.
Thinking of using heartbeat or something.
VRRP, Virtual Redundancy Router Protocol, an option?
Stefan de Konink
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Re: [Asterisk-Users] Door buzzer.

[Stefan de Konink]

On Sat, 4 Dec 2004, Cian O'Sullivan wrote:
> 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 the device is only a buzzer, can't you do anything fancy on the
comport, with hardware and an event poll?

Or if it is a phone device maybe an Iaxy can do the trick?

Stefan de Konink

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Re: [Asterisk-Users] Receptionist Phone

[Stefan de Konink]

Cian O'Sullivan wrote:
I have a customer interested in an * system, however she wants to ensure 
that the receptionist phone will display who is on the phone and who is 
not.  It is an office of 10 people, and there are 12 PRI channels available.

She is an older lady and does not want to use a web interface.  Any 
suggestions?
Using a Cisco with a XML browser and a CGI generated image, of who is on 
the phone at that time. Probably enough space to fit 10 persons in with 
a shrunk down font.

Stefan de Konink
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Re: [Asterisk-Users] Passing Var to PHP-AGI

[Stefan de Konink]

You are able to read the envirionment variables with the AGI command 'GET
VARIABLE '.

http://home.cogeco.ca/~camstuff/agi.html#GETVARIABLE
http://www.voip-info.org/tiki-index.php?page=Asterisk%20AGI
http://www.voip-info.org/wiki-Asterisk+AGI+php


Stefan

On Tue, 30 Nov 2004, Mike Roberts wrote:

> exten => auth_dial,1,DigitTimeout,5
> exten => auth_dial,2,ResponseTimeout,15
> exten => auth_dial,3,Read(dialed,IVR/en_enter_destination,0)
> exten => auth_dial,4,agi(call_start.php|${dialed})
> exten => auth_dial,5,dial(SIP/[EMAIL PROTECTED])
>
> I'm trying to get What they dialed put into the PHP script. How do I
> get the contents of this variable in the php script?
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Re: [Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)

[Stefan de Konink]

Gregory Junker wrote:
So, if anyone is interested, I am suggesting particularly a standalone, 
cross-platform project that is simple to install, configure, operate and 
manage. It should operate with or without a database. It can leverage 
existing projects, but it must not have the existence or installation of 
those projects as prerequisite. In other words, if this project uses 
another project's code, it must also include the installation and 
configuration of that project in this one's installer.
Again in this thread:
I thinking about a XUL application, which can be installed as a 
Mozilla/Firefox extension which connects to Asterisk?
Personally I presume this is not limited to Asterisk only management 
interface nor limited to Mozilla, but a standard framework which can be 
used to control application with, that support a Management Interface.

Stefan
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Re: [Asterisk-Users] $200 AMP documentation bounty < - Comments on the Linux user experience

[Stefan de Konink]

Colin Anderson wrote:
Again, it's good because programmers are motivated by writing cool code and
not concerning themselves with trivial things like documentation and UI. 
About cool code, why not make a Firefox Extension, which can connect to 
Asterisk. It is portable by XUL/Js and thus usable where ever Gecko 
runs, it can be run webbased but also stand alone. And probably a 
semi-portable XUL Phone can be made.

my euro 2 cents
Stefan
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Re: [Asterisk-Users] Gentoo

[Stefan de Konink]

Ed Brady wrote:
The latest portage tree has the latest release of  *.  However if you 
plan on keeping up to date with CVS head, I suggest you for-go using the 
portgage install, and use the source instead.
Or make a portage_overlay with an asterisk_cvs ebuild :)
Stefan
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Re: [Asterisk-Users] Gentoo

[Stefan de Konink]

Jay Milk wrote:
I have *read* that Gentoo would be a good distro for Asterisk or any
other servers which require optimal kernels and stable operation.  I
also know for a fact that some users here are running Asterisk on
Gentoo.  I have attempted several Gentoo installs (one Phase-1, one
Phase-2, and three Phase-3) last month, but have not been successful in
creating an operational server.
All the servers I've installed from phase - 1, mainly mail/webservers, 
run bright and shiny with linux2.6/glibc-nptl and that kind of stuff. 
The only problem that will happen with installing Asterisk is the __user 
cumma in ixjuser.h.

So if you post your problem here or on the gentoo forums/buzilla you 
probably get the help you need.

Stefan
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Re: [Asterisk-Users] How useful is the screen on IP phones?

[Stefan de Konink]

Kevin P. Fleming wrote:
More useful? No. More easily deployed and managed? By far, IMO.
If there was a stylish phone for a Grandstream price without screen 
would you choose a Cisco phone with screen?
I only would consider it if the phone with screen was better scriptable, 
but in a SIP Cisco image even the minibrowser is buggy. Last time i 
tryed (7960G pre 7) I needed to make a .cgi extension because a .php 
extension caused the phone to die.

Stefan
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Re: [Asterisk-Users] How useful is the screen on IP phones?

[Stefan de Konink]

Jay Milk wrote:
Thanks for the reply -- how well is this documented?  Is information
available from Cisco to endusers, or is this a big-money affair only?
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx
http://www.voip-info.org/wiki-Asterisk+Cisco+79XX+XML+Services
Stefan
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Re: [Asterisk-Users] How useful is the screen on IP phones?

[Stefan de Konink]

Kevin P. Fleming wrote:
João Amaro wrote:
I'm using 14 cisco 7940 as Dynamic queue agents.
They use the pixel based screen, to login/logout from queues. They can
also see the queues stats.
Now that's really not fair, to post a message like this without links to 
the code and/or documentation :-(

Can we assume from your message that you have implemented this yourself 
but are not making it available to the community?
Probably not using the SIP image too ;) Because with SIP you can only 
GET pages not PUSHing them to the user :)

But the other Cisco images provide some usefull coding abilities, but 
then again, is it more usefull then a PC with an integrated softclient 
(in the target software) + headset? I personaly don't think so.

Stefan de Konink
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Re: [Asterisk-Users] New Channel Driver: chan_bluetooth

[Stefan de Konink]

Hi,
As you may have noticed I also started working on a bluetooth channel, 
but since you already have a channel you probably are going to win this 
contest anyway.

I just tested the code in Asterisk 1.0.0 with a Nokia 6210i and a Jabra 
both connect over RFCOMM and send debug AT messages, but with neighter 
of them am I able to get any sound. Or make a call out to the phone.
I don't know if your code is having the same 'strange' sco timeouts as 
my test code has but I was pretty sure the Jabra-headset itself worked 
without those timeout.

An extension has been made to BLT/Jabra, but when calling to it nothing 
happens. I would be happy to help debugging and/or enhancing the code :)


Stefan de Konink
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Re: [Asterisk-Users] Re: GSM to g729 Conversion

[Stefan de Konink]

On Tue, 19 Oct 2004, Steve Kann wrote:
> Stefan de Konink wrote:
> > Isn't this an opportunity for Digium to offer encoded G729 files for a
> > fixed price directly encoded from the original wav files?
>
> I think this is an opportunity for people to use unencumbered codecs..
>
> If even just the asterisk community got  together to put half their G729
> money into speex enhancements, we'd all be much better off than lining
> sipro's pockets :)
(As mentioned before in the G723 discussion... I'm totally into Speex :)

But this was more a statement because this topic already popped up at
least 4 times in the last 15000 messages from which I can remember it.

It is not bad to prevent transcoding, the last time someone offered the
WAV files from Allison some users allready transcoded into other formats.
Why not make your files .speex too?


Stefan

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Re: [Asterisk-Users] Re: GSM to g729 Conversion

[Stefan de Konink]

Steve Underwood wrote:
- Original Message -
*From:* Matthew Boehm <mailto:[EMAIL PROTECTED]>
*Newsgroups:* gmane.comp.telephony.pbx.asterisk.user
*Sent:* Tuesday, October 19, 2004 6:03 PM
*Subject:* Re: GSM to g729 Conversion
There is no way to convert existing files to g729? The only reason
we need
the licenses is to access voicemail since they are in GSM.  All
our phones
have g729 built in. But if you try and access VM, you get that "No
coversion
for GSM to g729" error. But if all the voicemail sounds where in
g729, then
we don't need the licenses.
Matthew
Isn't this an opportunity for Digium to offer encoded G729 files for a 
fixed price directly encoded from the original wav files?

Greetings,
Stefan de Konink
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Re: [Asterisk-Users] GPL Violations (Was: Advice on OS Choice)

[Stefan de Konink]

On Thu, 14 Oct 2004, Brian West wrote:

> The GPL is still untested in the court system.

It stands in Germany now, Netfilter vs Sitecom GmbH, that is one of the
reasons why OpenXchange got Open Source & GPL.


Stefan de Konink

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Re: [Asterisk-Users] Web stream from an extension?

[Stefan de Konink]

Matt G wrote:
A EAGI interface with FFMPEG or Helix (using Windows 
Media/mplayer/Realplay as output), or if you are really good in 
Macromedia Flash directly stream it into the clients Flash player.

Stefan de Konink
what about a combo of an EAGI and MING ?
(generate flash on the fly)
ming.sourceforge.net
If have in depth experience with Ming, but the on the fly part is 
totally not related with this application. Ming indeed offers a way to 
generate Flash-on-the-Fly, but all that you want is a Flash application 
which is capable to connect to a stream (done by ActionScript) and play 
it. The only reason I can think of to (ab)use (fotf) Ming for this is 
security-by-obscurity for embedding the stream URL inside the 
application itself, instead of passing it as parameter or XML (session) 
request.

Making a Flash application in Ming isn't as easy as the Macromedia Flash 
Designer in the end it spits out roughtly the same code. And thus your 
choice which develop method you use for it. I do prefer Ming if it isn't 
design related but never use it for the On-the-Fly use.

(Compare it with an remote XUL application with can be made on the fly 
with PHP or the proper way, a XUL template and a PHP generated RDF file.)

Stefan de Konink
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Re: [Asterisk-Users] Web stream from an extension?

[Stefan de Konink]

Tony Mountifield wrote:
I want to create a CGI or PHP script that, when invoked from the web,
will make a call to Asterisk on a given extension number, and the audio
that is played to that extension gets streamed to the remote client.
A EAGI interface with FFMPEG or Helix (using Windows 
Media/mplayer/Realplay as output), or if you are really good in 
Macromedia Flash directly stream it into the clients Flash player.

Stefan de Konink
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Re: [Asterisk-Users] Embedded Asterisk System (was QoS Router/SoftwareSuggestions)

[Stefan de Konink]

Benjamin on Asterisk Mailing Lists wrote:
Use a dual CF adapter with two CF cards, mount one read-only for the
OS, Asterisk and drivers, mount the other read-write for /var/log and
voicemail.
Why can't you use a ramdisk and sync to CF on exit, I agree you do need 
a UPS for it...

Stefan de Konink
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Re: [Asterisk-Users] Bluetooth Bounty

[Stefan de Konink]

Jon Radon wrote:
Thanks for bringing this up again Jay.. I wonder how the people
working on the code are doing.. if they've had the time.
The Update:
At the moment we have testapplication connectivity with the Nokia 6310i 
and the Jabra headset. With the side note that this connectivity for the 
6310i timeouts (connection reset by peer).
Together with Nate, I am trying to get his Ericson working because his 
phone times out even faster then my Nokia.

So atm we are debugging...
Stefan de Konink
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Re: [Asterisk-Users] What about a higher level configuration language

[Stefan de Konink]

Brian Capouch wrote:
Of course, the problem of the hard-coded "priority + 101" situation is 
problematical.  I say we think through what the perfect world would look 
like in this respect and then see how hard it would be to implement. . .
XML will probably able to store much, probably more, of the flat text in 
a marked up as 'Asterisk XML', created with a nice XSL-template there 
would even for +101 situations not be any problems.

But my major consern is the validness of goto statements, but thats 
probably also the issue with the current 'list' versions.

It is allready possible to start developing an 'Asterisk XML' since with 
an XSLT preprocessor you are able to generate a valid extension.conf.


Stefan de Konink
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Re: [Asterisk-Users] GSM phones, bluetooth and general happiness

[Stefan de Konink]

Jay Milk wrote:
That's exactly it!  The asterisk box acts as a handset for the phone and
uses AT-commands for call-origination and progress.
OFF: Although the reversed thing, having a console with a bluetooth 
headset would also sounds very ok.

ON: Why would one prefer bluetooth over wires if one still needs a 
serial cable to send AT commands? For an incomming call, GSM to 
GSM-Asterisk it would make sense, but I never saw a bluetooth headset 
device with keypad. Probably not only a bluetooth API should be looked 
at moreover the to be used Cellphones API.

Stefan de Konink
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Re: [Asterisk-Users] Cheapest SIP Phone

[Stefan de Konink]

SeshKanuri wrote:
We want  to beat grandstream both at features and price.
I can sell these industry standard PA1688 Chip enabled phones with IAX2, yes
I said IAX2 (along with SIP, H323 and MGCP and a few more such protocols
already enabled) at bulk rates to anyone interested in them.
So does that also includes the phones firmware sources?
Stefan de Konink
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Re: [Asterisk-Users] Re: X100p on VIA EPIA-V

[Stefan de Konink]

Rich Adamson wrote:
My understanding from the 'expert' is the PCI issues have something
to do with a poor PCI chip design on the motherboard. The folks that
are heavy into audio apps tend to swap out their VIA motherboards. 
Guess that implies there aren't any workarounds.
From the Ardour (Linux Audio Editor) System requirements:
Avoid VIA motherboards and chipsets wherever possible. This company has 
demonstrated an almost complete disregard for reasonable use of the PCI 
bus. Their hardware has repeatedly been implicated in a failure to 
achieve low latency performance. Here is one example of the kinds of 
problems you can expect.

example points to:
http://linux.derkeiler.com/Mailing-Lists/Kernel/2003-09/7761.html

Stefan de Konink
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Re: [Asterisk-Users] Free WWT (WorldWideTelco): Utopia, or just a matter of organization?

[Stefan de Konink]

Hi,

Without going in depth of my thesis research, this is _no_ utopia and
with a good organisation it can be set very easily also by non techies.

Probably everybody remembers Sipphone.com with there *cute* little
blackboxes where you can put in a landline, ethernet, and a regular phone
(pots). With the thingie you can switch landline/vip by placing a * before
a phonecall. Rest can be setup in a webinterface.

This box is stupid but cheap! So what is my proposal to make it smart?
Implementing ENUM in it. Something close to the DNS/ITU version,
but with this difference that least cost routing is in place. Basically
the SIP proxy/Asterisk where you connect to feets the DNS database with
replacements for tel:phonenumber to sip:[EMAIL PROTECTED]

DNS (ENUM) knows when the user calls to a non-voipable localcall, thus,
the tel:localnumer would tranfer to the real phonenet. [*]

But... since I cannot get any hold of Leadtech and the company who
designed this box refers to them I cannot get this idea into practice. But
as you know this box is only a FXO/FXS with Eth0, things Asterisk has got
too, by implementing this smart routing in a dialplan format (Dynamic ENUM
is this case) your utopia gets a GPS position on the map.


[*] I'm, at the moment, trying to extend DNS with some privacy aspects
making ENUM for myself more valuable. With this same privacy extensions it
would be possible to _identify_ a user and generate the right responce.


This all sounds great if you have a userbase, without it I could only call
only call yourself on my brand new communication system...



Greetings,

Stefan de Konink

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Re: [Asterisk-Users] Commercial CID spoofing system

[Stefan de Konink]

Joshua M. Thompson wrote:
Looks like an Asterisk box and a simple CGI script to me.
Is this possible out there without a SS7 gateway? Or do you need just a 
friendly channel supplyer that allows you to send any callerids thru 
their switches?

Stefan de Konink
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Re: [Asterisk-Users] Commercial CID spoofing system

[Stefan de Konink]

Brian Capouch wrote:
FYI.  Reading is free; if you don't have an account it is trivial to 
sign up, and they're very politically correct, as might be imagined, 
about using email for selling purposes.

http://www.nytimes.com/2004/09/02/technology/02caller.html?hp
Bugmenot.com:
Login details for www.nytimes.com
Account #1
xxgeo
12345
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Re: [Asterisk-Users] Strange behaviour using 7960

[Stefan de Konink]

On Tue, 20 Jul 2004, Brian D'Arcy wrote:

> Anyone ever seen anything like this before?

Yes, with a Grandstream over an ADSL (routed) line. I disabled the check,
but the problem stil occured. And since this friends line was almost the
lowest ping in the complete ISP network something else must be responsible
for Asterisk thinking:  :o bad line :o

At my own place I have 7960 and ADSL (PPTP) line but I connect to
Asterisk via a IPIP Tunnel. In the beginning I had the opposite problem, I
could hear the other party but they could not hear me.
But I don't remember having this problem myself in the last months
anymore.

Stefan

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[Asterisk-Users] AGI Dial, Extension dial SIP Loop

[Stefan de Konink]

At the moment I'm prototyping an advanced ENUM application with PHP
fetched from LDAP. When a user enters a full hostname as SIP adress I get
loop problems from the AGI EXECUTE DIAL and from a Dial in the
extension.conf.

-- Executing AGI("SIP/1000-c3c3", "enum.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/enum.php
  enum.php: 123
  enum.php: 3170327
  enum.php: LDAP bind successful...
  enum.php: telephoneNumber=3170327,ou=People,dc=eshara
  enum.php: sip:[EMAIL PROTECTED]
  enum.php: in: sip:[EMAIL PROTECTED]
  enum.php: uit: sip/[EMAIL PROTECTED]
  enum.php: in: sip:[EMAIL PROTECTED]
  enum.php: uit: sip/[EMAIL PROTECTED]
-- AGI Script enum.php completed, returning 0
-- Executing Dial("SIP/1000-c3c3", "sip/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 482 "Loop Detected" back from xxx.xxx.xxx.xxx
  == No one is available to answer at this time
-- Executing Hangup("SIP/1000-c3c3", "") in new stack
  == Spawn extension (default, 3170327, 3) exited non-zero on
'SIP/1000-c3c3'


But when I skip the @asterisk.blabla.bla it strangely works from the
extension.conf but not from the AGI script directly.

Now I set a variable, and then call do a:

AGI:
 write("SET VARIABLE CALLTHIS ".uri2tech($info[0]['description'][0]));
Extension:
 Dial(${CALLTHIS})

-- AGI Script enum.php completed, returning 0
-- Executing Dial("SIP/1000-a5c0", "sip/1000") in new stack
-- Called 1000
-- SIP/1000-d8a9 is ringing
  == Spawn extension (default, 3170327, 2) exited non-zero on
'SIP/1000-a5c0'


I want to know why it fails with:
write("EXEC Dial ".uri2tech($info[0]['description'][0]));

Is there a way to get this to work without stripping the hostname part?
How did other users solve this problem while using ENUM as backend and
calling locally?


Greetings,

Stefan de Konink

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[Asterisk-Users] Directed Call Pickup

[Stefan de Konink]

In the list I found some messages that *8 doesn't work so well. Is there 
any possibility to create a extention that you can call, and if you are 
fast enough, pick up a number? (Also if you are outside your callgroup)

like
pseudo code:
exten => 888, 1, EnterPhoneNumber()
exten => 888, 2, EnterPass()
exten => 888, 3, TransferCallToThisPhone()
exten => 888, 103, Invalid()
I'm interested in how to do the 'TransferCallToThisPhone' procedure.
Stefan
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RE: [Asterisk-Users] VoIP hackers gut Caller ID

[Stefan de Konink]

MCI definately does this. We tryed out a sample to replace our CallerID
with the one we forwarded. Did not work :( otherwise it was really cool.

But I can imagine if someone talks SS7 noone could 'touch' them, or isn't
that true?


Stefan

On Wed, 7 Jul 2004, Kevin Walsh wrote:

> Adam Hart [EMAIL PROTECTED] wrote:
> > Chris Foster wrote:
> > > The Register is carrying a article written by Kevin Poulsen of
> > > Securtiy Focus, calling asterisk  "..the most powerful tool for
> > > manipulating and accessing CPN data.."
> > >
> > > I hope NuFone doesn't drop asterisk-set-able callerid's after this
> > > article; i've been wanting that feature from voicepluse for a long
> > > time.
> > >
> > These kind of things will be reason (excuse) for Voip to be regulated
> >
> Perhaps service providers who allow the Caller*ID to be set should
> insist that customers provide evidence that they own the phone numbers
> that they want to publish, and then limit the customers' choices to
> only the numbers in their approved list.  Calling the customer on the
> provided number(s) would be an easy way to check, and a setup fee
> could be levied to cover the provider's time and expenses, if required.
>
> Being able to discover a "blocked" Caller*ID is another matter.  Both
> are good areas for regulation.
>
> --
>_/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
>   _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
>  _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
> _/   _/  _/_/_/_/  _/_/_/_/  _/_/
>
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Re: [Asterisk-Users] Special Delivery from China

[Stefan de Konink]

Is it possible to ask this 'friends' to incorperate the Speex codec in 
the phone. That would be like a real cool feature.

Stefan
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Re: R: [Asterisk-Users] How to force G729

[Stefan de Konink]

When I set the SIP_CODEC variable to force g729:

Jun 24 12:30:01 NOTICE[1226062640]: chan_sip.c:1313 sip_answer: Changing
codec to 'g729' for this call because of ${SIP_CODEC) variable
-- Attempting native bridge of SIP/8011-86fe and SIP/8008-c2b9
Jun 24 12:30:02 NOTICE[1234455344]: channel.c:1508 ast_set_read_format:
Unable to find a path from G729A to ULAW
Jun 24 12:30:02 NOTICE[1234455344]: channel.c:1478 ast_set_write_format:
Unable to find a path from ULAW to G729A
Jun 24 12:30:02 WARNING[1226062640]: chan_sip.c:1332 sip_write: Asked to
transmit frame type 256, while native formats is 4 (read/write = 4/4)
  == Spawn extension (sip, 8041, 2) exited non-zero on 'SIP/8011-86fe'
-- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back
from 217.117.xxx.xxx

Though I get a short 'hello' (voice) from the otherside, but after that
line dies.

Stefan

On Thu, 24 Jun 2004, Manuel Wenger wrote:

> >Try to configure in sip.conf your extensions context like this:
> >
> >[XXX]
> >
> >disallow=all
> >allow=g729
> >
>
>
> Done that already: but then, the "incoming channel" (from the user to Asterisk) is 
> G729, and the "outgoing channel" (from Asterisk to the PSTN gateway) still remains 
> ULAW, so Asterisk has to do transcoding (ULAW to G729), which I don't want, 
> obviously.
>
> For some reason, the outgoing channel doesn't follow the setvar(SIP_CODEC=g729) rule.
>
> -Manuel
>
>
> ___
> Ticinocom SA - Via Stazione 5 - 6600 Muralto
> Tel 0844 007070 - Fax 0844 007071
> http://www.ticinocom.com
>
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[Asterisk-Users] Skype 4 Linux

[Stefan de Konink]

Hi All,

Since 21 june skype is available to be used on Linux, with a static
binary, which includes QT, of 8 meg its big.

http://www.skype.com/help_linux_faq.html

I presume, with some hacking, there could be a possibility to use the
Skype program as a Channel. (Eq. Skype is started, and with a visual
scripting thing a connection is made and Asterisk connects via OSS (or the
alsa emulation layer)).

It is a bit of work, but reverse enginering is too :)



Stefan

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RE: [Asterisk-Users] LDAP synchronization script

[Stefan de Konink]

The base problem, I presume is not that there is no documentation, but how
to combine all those defacto standards, from an user and an application
point of view.
An Active Directory implementation in Linux (for users and application)
for me starts with the standard PAM/NSS stuff but why not extend that for
Jabber, Asterisk, Postfix/Sendmail, DHCPd, DNS and a zillion other stuff
like (a higher level) ENUM?

For most of the above application are 'dynamic' ldap backends made, which
are usable. Though what is the best thing to start with? Application with
users under it. Users with Application under it. Or the last type I think
it is the most usuable way of implementing:

Organisation/
Groups/
Applications(Group/Application Specific configs)
Users/
Applications(User/Application Specific configs)
Applications(Organisation Specific configs)

Applications (Basic configuration)
Name/   (Name like Asterisk)
ID/ (Which Asterisk server IP address etc.)


Which makes .application and /etc/application obsolete if well
implemented. Performance wise you would not want to poll the LDAP server
24/7 (though I want it ;) but only fetch while reloading.

In the combination and integration of those things I'm now writing a
thesis with a production proof-proof of concept, for Unified Messaging in
a Box. Though, importing all schema's like cosine, dhcpd, etc. the mess
only gets bigger eq. there need to be a basic structure and I would
like to have some feedback about it.

The main objective is to make the user have a 'home' peer/server, though
it doesn't depends on this peer but it is like 'the first choice'. For
example two Asterisk servers, one crashes the other peer/server takes over
and starts accepting the other servers its users.

Ok, this basically implies there is a distributed filesystem around, at
the moment I use CodaFS for that. (Requires patching of some programs like
Postfix)


Stefan


On Fri, 18 Jun 2004, Lars Boegild Thomsen wrote:

> Hi,
>
> > The what belongs were is my big question at the moment and I personally
> > don't want to design anything LDAP-ish that would become my private tree
> > instead of defacto implementation.
>
> You should definitely have a look at the defacto standards for storing users
> and groups (check http://www.padl.com/OSS/pam_ldap.html).  Would be rather
> cool to have a Linux network with users and groups defined in LDAP - and
> each user just having an extension defined in his record.  Asterisk base
> configuration should go in separate three.
>
> Regards,
>
>   Lars...
>
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Re: [Asterisk-Users] Asterisk as Internet Talk Radio PBX system

[Stefan de Konink]

To use Asterisk as platform for such a system you probably want to have a
Alsa enabled card which supports routing of multiple channels in and out.
So Asterisk is like the intermediate 'engine' that routes the signal. (Or
sort some soft-mixer). A user is then placed in a Meetme room and the
hold signal would be the live show?

OUTPUT||==o
  ||
Soundcard 1 (studio/broadcast)   =||=|---\
 ||
Soundcard 2 (prelisten/desk) |\   |
 | |  |
SIP/IAX incomming|/   |
 ||
Meetme   |---/
 |
Meetme   |
 |
Meetme   |

Technical picks of the phone by prelistening, transfers it to a new or
existing meetme. When the actual 'meeting' starts, the meetme gets in the
air studio audio (presenter) gets in by the Alsa interface.

Somebody earlier suggested Asterisk for use in remote broadcasts (on a
location for example). With two boxes and a ISDN line on both sides, some
encoding and you are in business too.

Asterisk as a application platform is quite powerfull, but probably has
some overhead which 'all-in-one' products have.


Stefan

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RE: [Asterisk-Users] LDAP synchronization script

[Stefan de Konink]

Yeah that 'old' message discribes VERY MUCH what I'm doing at the moment.
Though there should be an 'application' part and an universal 'user' part.

For example the meetme is application specific, should be in the Asterisk
tree. But the extentions should basically be templates part of the
Asterisk tree which can be used in the universal 'user' part.

The what belongs were is my big question at the moment and I personally
don't want to design anything LDAP-ish that would become my private tree
instead of defacto implementation.

Stefan

On Thu, 17 Jun 2004, David Hajek wrote:

> I think I'll use something from this article -
> http://www.marko.net/asterisk/archives/0205/0006.html
>
> -David
>
> > -Original Message-
> > From: Stefan de Konink [mailto:[EMAIL PROTECTED]
> > Sent: Thursday, June 17, 2004 1:12 PM
> > To: David Hajek
> > Cc: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] LDAP synchronization script
> >
> > I'm planning to incorporate this (native and dynamic) LDAP
> > for my own system on short term. Do you have any LDAP design in mind?
> >
> > Stefan
> >
> > On Thu, 17 Jun 2004, Jeremy Jones wrote:
> >
> > >
> > > > David Hajek
> > > > Sent: Thursday, June 17, 2004 2:41 AM
> > > > To: [EMAIL PROTECTED]
> > > > Subject: [Asterisk-Users] LDAP synchronization script
> > > >
> > > > Hello,
> > > >
> > > > I understand there's no possibility to have asterisk
> > configuration
> > > > (sipusers, extensions, voicemail) in LDAP right now. I'm thinking
> > > > about put the (sipusers, extensions, voicemail) info in LDAP and
> > > > then run some synchronization script on the asterisk server which
> > > > will build up appropriate configuration files and reload asterisk.
> > > >
> > > > I'm sure this script is already around. Can some share
> > one with me/us?
> > > >
> > >
> > > Not aware of any scripts like that, but...
> > > you could use the odbc support in asterisk in conjunction with some
> > > slick odbc-ldap connectivity.
> > >
> > > Jeremy Jones
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> >
> >
> >
>
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Re: [Asterisk-Users] embedded Asterisk

[Stefan de Konink]

On Thu, 17 Jun 2004, listas iPfone wrote:

> 1. burn the rescue iso
mount -o loop -t iso9660 /file /mnt/loop

> 1. copy the rescue disk to a hard drive
cp -dpR /mnt/loop/* /new/location

> 2. compile asterisk
make PREFIX=/new/location install (check if asterisk don't copy all
development non-sence)

> 3. copy all to the flash disk
fdisk /dev/hdX[0-9]
make partitions
mkfs.ext2 /dev/hdX[0-9]
mount -t ext2 /dev/hdX[0-9] /mnt/flash
cp -dpR /new/location /mnt/flash


> It is that simple?
Probably you want something that actually boots the system too. I don't
know if the ISOLINUX pakage supports a LILO kind of thing, but I guess it
does. That should be in the MBR of your flash disk and you could probably
boot it. I wrote the instructions by mind, so probably something is
missing :)

Stefan

>
> Miklos
>
> - Original Message -
> From: "Klaus-Peter Junghanns" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, June 17, 2004 5:11 AM
> Subject: Re: [Asterisk-Users] embedded Asterisk
>
>
> > Hi,
> >
> > > Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at
> > > 233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which
> > > is a downstripped Debian (< 64 MB) on a readonly ext2 filesystem, you
> > > should be grand. Installing asterisk + some extra stuff will probably
> > > require, that you have at least a 128MB or 256MB flash or so.
> >
> > Dont go for "stripped down but complete" distributions which include a
> > lot of stuff that you dont need, e.g. gcc. Go for a rescue system, like
> > i used the SuSE rescue system (14 mb), then you can add what you need
> > (sshd,...) and compile asterisk on another box and then just copy it.
> > My compressed ramdisk image is 32 mb, including all voice prompts and
> > some mp3s for MOH.
> >
> > >
> > > There are actually quite some board around on that CPU, like Soekris,
> > > pcengines and i think also Mikrotik at prices from 120EUR and up.
> > >
> > I just put together the demo system for Linuxtag:
> > - Via EPIA 5000 (C3-533), EUR 80,-
> > - Morex case with external power supply, EUR 80,-
> > - some old 256 mb SDRAMM
> > - 128 MB USB memory stick, EUR 30,-
> > - 1 quadBRI (could also easily handle an octoBRI, or a PRI card,
> >   with the dual riser pci card you can use 2 cards)
> >
> > The C3-533 is an i586 CPU. According to "show translation" it needs
> > 30 ms for transcoding 1 channel from g711 to gsm (and vice versa).
> > So, neglecting any overhead caused by channel handling it could
> > transcode 30 channels to gsm.
> >
> > Linux BIOS has support for the EPIA boards, so you can speed up booting
> > very much and also disable the VGA port (very useful for production
> > deployments).
> >
> > > I'm running pebble on a pcengines board, just needed to customize the
> > > kernel a bit, haven't been testing asterisk on that yet, but i definatly
> > > will in the sooner future.
> > >
> > > Kind regards,
> > > Martin List-Petersen
> > > martin (at) list (dash) petersen (dot) net
> >
> > best regards
> >
> > Klaus
> > --
> > Klaus-Peter Junghanns
> >
> > CEO, CTO
> > Junghanns.NET GmbH
> > Breite Strasse 13a - 12167 Berlin - Germany
> > fon: (de) +49 30 79705390
> > fon: (uk) +44 870 1244692
> > fax: (de) +49 30 79705391
> > iaxtel: 1-700-157-8753
> > http://www.Junghanns.NET/asterisk/
> >
> >
> > ___
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> >
>
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RE: [Asterisk-Users] LDAP synchronization script

[Stefan de Konink]

I'm planning to incorporate this (native and dynamic) LDAP for my own
system on short term. Do you have any LDAP design in mind?

Stefan

On Thu, 17 Jun 2004, Jeremy Jones wrote:

>
> > David Hajek
> > Sent: Thursday, June 17, 2004 2:41 AM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] LDAP synchronization script
> >
> > Hello,
> >
> > I understand there's no possibility to have asterisk configuration
> > (sipusers, extensions, voicemail) in LDAP right now. I'm thinking
> > about put the (sipusers, extensions, voicemail) info in LDAP
> > and then run
> > some synchronization script on the asterisk server which will build up
> > appropriate configuration files and reload asterisk.
> >
> > I'm sure this script is already around. Can some share one with me/us?
> >
>
> Not aware of any scripts like that, but...
> you could use the odbc support in asterisk in conjunction with some
> slick odbc-ldap connectivity.
>
> Jeremy Jones
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>


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Re: [Asterisk-Users] embedded Asterisk

[Stefan de Konink]

Probably the best thing to do is to build a uClibc tree, disable some 
Asterisk codecs (which don't want to compile, first run) compile again 
and run.

Tomorrow I'm going to do the samething for an Epia-MII 
1,2GHz/512MB/512MB-CF. Another tip :P Don't compile on flash... just 
make a tree on your harddisk. And copy the required binaries and libs to 
a root tree and attach a kernel. Look at some different Filesystems too, 
depending on for needs Ext2/Minix/CramFS.

Btw. for what purpose do you want to run the box? I can imagine that a 
few voicemail messages can float the system. And if SIP is only required 
you should probably use SER for the project. I want to try out the VOCAL 
footprint too but didn't had the time to do that yet.

Stefan
listas iPfone wrote:
Hi All,
 
I have a thin cliente here that i want to run asterisk:
 
- National Semicondudor Geode GX1 266MHz Geode 266MHz single chip 

-  NS Cx5530a Southbridge National Semiconductors SC2200 

 - NS PC97317 in chipset 

 -  32MB Compact Flash 
 - 64MB Ram 
 
- 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National 
DP83815 / DP83816

Some tip?
 
I have a ide>flash adaptor to make the install...
 
I need recomendations in Linux distro... asterisk min. install 
...etc..any info i can get.
 
Thanks for any help
 
Miklos

Atenciosamente
Cláudio Miklos
* iP FONE *Telefonia IP
Rua Caio Graco 735 São Paulo SP
( BR - 55 11 3801-3702
( USA - 1 360-968-1591
( FWD - 64662
( sip:[EMAIL PROTECTED]
www.ipfone.com.br 
[EMAIL PROTECTED] 
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Re: [Asterisk-Users] limitations ?

[Stefan de Konink]

If you take a look at http://sipp.sourceforge.net/ there is a utility 
which claim to check the SIP performance of the specific system. (btw 
don't try this on a target number which has voicemail, then the test 
becomes a bit subjective ;)

I see asterisk more and more as real cool pbx with features instead of a 
dumb switching board. If someone can tell when * shouldn't be used as a 
single box solution, but as a group of products (SER, Vocal, Asterisk, 
Gatekeeper etc.) and print that in a chart would be good for the VoIP 
industry.

Stefan
Harold Workman wrote:
hi,
im looking at deploying asterisk in a small corporate enviroment which will
have approx. 1200 IP Phones and an average of about 100 to 200 calls at any
given time.   The calls will be sent out  SIP to my Cisco Gateway.  Im
running Asterisk on a Dell Dual P3 1.2ghz running Fedora.   Is there a
calculator or a spreadsheet which could tell me about how many calls I will
be able to make through * ?
---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
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[Asterisk-Users] SIP->Application Codec debugging

[Stefan de Konink]

Hello,

After testing some different phones, codecs and combinations of them I
noticed that some of my GSM applications didn't work anymore. So after
finding out what it was (no codec support) I was thinking, why * couldn't
give my that error directly (-d -vvvr did not give any feedback). One
thing for sure, it didn't do any codec translation or gave an error about
that.

An error if -> an application <- can't agree upon codec would be nice, is
there allready a way to see this 'newbie' SIP errors?


Stefan

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Re: [Asterisk-Users] XML How To for Cisco 7960

[Stefan de Konink]

http://ipphones.utelisys.net/
http://ipphones.utelisys.net/includes/cisco.inc.phps

There are some perl classes on this topic too (even for image
generation!). I didn't had the time to made a GD patch to use it inside
PHP yet. But I hope this wil help. Anyway on Cisco.com you can find some
PDF files with clear statements. Only thing that doesn't work is HTTP_PUSH
:(


Stefan

On Thu, 10 Jun 2004, Matthew John Darnell wrote:

> Aloha,
>
> Has anyone written an XML application for a Ciso 7960 phone running SIP?
>
> I can't find any examples anywhere!
>
> Anyone know of any resources for this?  I have read it can render XML & can
> get input from the keypad & softkeys.
>
> Aloha,
> Matt
>
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Re: [Asterisk-Users] Cisco 7970 w/ 7.1 phones rebooting with asterisk

[Stefan de Konink]

Currently I'm using 6.2 which is obviously stable, though if you look at
the release notes on the Cisco site for 7.0/7.1

http://ftp-sj.cisco.com/cisco/voice/ip-phone/sip-7960/phnrn70s2.pdf
http://ftp-sj.cisco.com/cisco/voice/ip-phone/sip-7960/phnrn71s2.pdf

(You need to be loged in for that)

There are a lot Cavecats. Are there any real new features in 6.3+ to
switch over (like HTTP_PUSH which probably is still not availeble for the
SIP series)?

Stefan

On Thu, 10 Jun 2004 [EMAIL PROTECTED] wrote:

> I am currently testing an asterisk server with some cisco 7960 phones.
> I have been having problems with phones rebootin using 6.3 firmware in
> the asterisk voicemail menus.  The phones reboot after a dozen or so
> random button presses while in the voicemail menus.  To try and fix
> this, I upgraded to sip 7.1 only to find that now the phones reboot
> even if i'm trying to press a button to dial out.  I am running CVS
> head as of 2 days ago.  If anyone has ever had any problems like this
> please let me know
>
> Nathan
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Re: [Asterisk-Users] Re: Compiling Asterisk with G.723.1

[Stefan de Konink]

http://asterisk.gnuinter.net/files/digium/asterisk-ng/db1-ast/

I 'stole' the sources from them, compiled (and working) after uncommenting
the makefile. Tobad only the Budgettones work with the codec and not the
Cisco's :(


Stefan

On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote:

> Hello,
>
> I have the whole library (G.723.1 and G.723.1b) downloaded from ITU, but
> it doesn't compile with Asterisk "out-of-the-box".
>
> So, unless someone else can provide a library which compiles with *,
> we'll have to tinker with the ITU source code (if it is possible at all).
>
> Best regards,
> Vlasis Hatzistavrou.
>
> Stefan de Konink wrote:
> > So simple question, without googling:
> >
> > Where can the g723.1 and g723.1a be found... so a 'EU-patch' can be make.
> > I'm able to host it in Amsterdam.
> >
> > Greetings,
> >
> > Stefan de Konink
> >
> > On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote:
> >
> >
> >>Randy Ackers wrote:
> >>
> >>>>Tony Hoyle wrote:
> >>>
> >>>
> >>>>Steve Underwood wrote:
> >>>>
> >>>>
> >>>>>I didn't say one patent covered all the world. I said the patents on
> >>>>>codecs exist all over the >>world. WIPO is simplifying this a bit,
> >>>>>but its still pretty expensive to get a patent everywhere. I >>know
> >>>>>of no country where the key aspects of a codec cannot be patented.
> >>>>>
> >>>>>Outside the US you can't patent software or algorythms, and a codec
> >>>>>is (usually) both of these, >>therefore not patentable outside the
> >>>>>US.  This is what allows things like the xvid project to exist, >>for
> >>>>>example, which breaks several US patents...  Fraunhoffer somehow
> >>>>>apparently managed to >>get some in europe but it was never decided
> >>>>>whether they were valid or not (commonly it is >>thought that they'd
> >>>>>have failed under legal challenge as the wording of EU patent law is
> >>>>>very >>clear).
> >>>
> >>>
> >>>>>Try looking up the EU patents related to any of the ETSI codecs, like
> >>>>>GSM EFR, half rate, AMR, >>etc. If Fraunhoffer's patents can be
> >>>>>challenged, they must have screwed up the way they >>worded them.
> >>>
> >>>
> >>>===
> >>>Hello,
> >>>
> >>>I think that the discussion has strayed from its original subject: the
> >>>subject is WHERE is the library for the G723.1 codec in Asterisk.
> >>>
> >>>There are many people/companies/organizations who need G723.1. Although
> >>>apparently it's not a problem using a patented codec like G723.1 outside
> >>>of the USA, most of us would gladly pay a reasonable per-channel fee for
> >>>it's usage, like in the case of the G729 which Digium offers.
> >>>
> >>>But since it is not available in this manner, I think it's only fair to
> >>>provide the source code for compilation/usage at least outside of the US.
> >>>
> >>>I know that quite a few Asterisk users have compiled G723.1 in their
> >>>box. Like many others, I would like to have this code and be able to
> >>>compile it in my box.
> >>>
> >>>In fact, many of us would even pay a reasonable sum in order to have the
> >>>code, if the people who already have it & use it in their boxes are not
> >>>willing to share for free.
> >>>
> >>>Regards,
> >>>Randy Ackers.
> >>>
> >>>_
> >>>MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*.
> >>>http://join.msn.com/?page=features/virus
> >>>
> >>>___
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> >>>
> >>
> >>I agree with Randy, G.723.1 would be extremely useful to many.
> >>
> >>And since G.723.1 could be used outside of the US from what I
> >>understand, it wou

Re: [Asterisk-Users] Re: Compiling Asterisk with G.723.1

[Stefan de Konink]

So simple question, without googling:

Where can the g723.1 and g723.1a be found... so a 'EU-patch' can be make.
I'm able to host it in Amsterdam.

Greetings,

Stefan de Konink

On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote:

> Randy Ackers wrote:
> >> Tony Hoyle wrote:
> >
> >
> >> Steve Underwood wrote:
> >>
> >>>>
> >>> I didn't say one patent covered all the world. I said the patents on
> >>> codecs exist all over the >>world. WIPO is simplifying this a bit,
> >>> but its still pretty expensive to get a patent everywhere. I >>know
> >>> of no country where the key aspects of a codec cannot be patented.
> >>>
> >>> Outside the US you can't patent software or algorythms, and a codec
> >>> is (usually) both of these, >>therefore not patentable outside the
> >>> US.  This is what allows things like the xvid project to exist, >>for
> >>> example, which breaks several US patents...  Fraunhoffer somehow
> >>> apparently managed to >>get some in europe but it was never decided
> >>> whether they were valid or not (commonly it is >>thought that they'd
> >>> have failed under legal challenge as the wording of EU patent law is
> >>> very >>clear).
> >
> >
> >>> Try looking up the EU patents related to any of the ETSI codecs, like
> >>> GSM EFR, half rate, AMR, >>etc. If Fraunhoffer's patents can be
> >>> challenged, they must have screwed up the way they >>worded them.
> >
> >
> > ===
> > Hello,
> >
> > I think that the discussion has strayed from its original subject: the
> > subject is WHERE is the library for the G723.1 codec in Asterisk.
> >
> > There are many people/companies/organizations who need G723.1. Although
> > apparently it's not a problem using a patented codec like G723.1 outside
> > of the USA, most of us would gladly pay a reasonable per-channel fee for
> > it's usage, like in the case of the G729 which Digium offers.
> >
> > But since it is not available in this manner, I think it's only fair to
> > provide the source code for compilation/usage at least outside of the US.
> >
> > I know that quite a few Asterisk users have compiled G723.1 in their
> > box. Like many others, I would like to have this code and be able to
> > compile it in my box.
> >
> > In fact, many of us would even pay a reasonable sum in order to have the
> > code, if the people who already have it & use it in their boxes are not
> > willing to share for free.
> >
> > Regards,
> > Randy Ackers.
> >
> > _
> > MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*.
> > http://join.msn.com/?page=features/virus
> >
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> I agree with Randy, G.723.1 would be extremely useful to many.
>
> And since G.723.1 could be used outside of the US from what I
> understand, it would be very practical if the source code was available
> for compilation & use on Asterisk.
>
> Thanks,
> Vlasis Hatzistavrou.
>
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Re: [Asterisk-Users] Asterisk Receptionist manager program.

[Stefan de Konink]

First feedback...

Bit (very) disappointed this is a Win32 program :( I had hoped to find
some crossplatform sources, but probably many other people like the idea.


Stefan

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Re: [Asterisk-Users] Router, Firewall, SIP Rewriter, and GnuGK

[Stefan de Konink]

Gentoo and emerging works well on new hardware (or with use of DistCC),
most of the applications direct from the portage (to be compiled
packages).

Stefan

On Tue, 1 Jun 2004, Serge Mankovski wrote:

> Hi
> I am running firewall/router "brew" made of RedHat, Shorewall, Siproxd and
> GnuGK on a box that connects through PPPoE to Internet. I run Asterisk on
> another box behind of it and it seem to work fine for me.
>
> I am thinking of replacing the router box because hardware is getting flaky.
> I do not want to go through pain of assembling all this stuff together
> again. Does anybody know of a Linux distro that would have all these things
> running in it "out of the box"? It does not need to have the same
> components, but should have the same function.
>
> Thank you
> Serge
>
> _
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Re: [Asterisk-Users] Hyperthreading?

[Stefan de Konink]

On Tue, 1 Jun 2004, Terry Goodwin wrote:

> BTW,  it seems the OS thinks there are 4 processors installed.  Even
> core 2 (2.6.5 kernel) when briefly installed (because it sucks) reported
> 4 CPU's.


Trust me... this is what you want :)

Stefan

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Re: [Asterisk-Users] Linksys

[Stefan de Konink]

On Mon, 31 May 2004, Greg Boehnlein wrote:

> On Mon, 31 May 2004, Tony Hoyle wrote:
>
> > Stefan de Konink wrote:
> > > Is Asterisk not a *little bit* too much for that processor? SER could be a
> > > better choice?
> >
> > The asterisk binary alone is larger than the total flash ram space on the linksys.
> >
> > I really doubt it's going to work
>
> That assumes that compiling for the MIPS w/ uClibc is going to result in a
> binary that is similar in size to that of an x86 binary, which isn't
> neccessarily going to be true.

For your information * compiles in a clean uClibc env. expect for one or
two codecs, who use some x86 specific assembly. I have it on my system for
testing with an Epia.

Ok... you could run it with a remote NFS mount :-)


Stefan

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Re: [Asterisk-Users] Linksys

[Stefan de Konink]

Is Asterisk not a *little bit* too much for that processor? SER could be a
better choice?

Stefan

On Mon, 31 May 2004, Girouard, Marc wrote:

> Wondering if anyone tried to port Asterisk to the Linksys 54G OpenSource
> platform?
>
>
>
> I am planning to try to port some of the Asterisk code to that platform and
> if any once already tried I would like to get in touch with them . I am
> thinking on porting the protocol and some other application but not the
> Codec itself.
>
>
>
> MarcG.
>
>
>
>

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Re: [Asterisk-Users] MacOS X softphone IAX clients?

[Stefan de Konink]

http://iaxclient.sourceforge.net/

On Sun, 30 May 2004, Tor Houghton wrote:

> Are there any softphone clients that can use IAX/IAX2 for MacOS X?
>
> Regards,
>
> Tor
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RE: [Asterisk-Users] Asterisk Receptionist manager program.

[Stefan de Konink]

> We are designing it for a touch screen monitor for her to do transfers,
> see whose on the phone and a few other features. Its in the development
> stage and has bugs.
> but I think its gonna be really good.
Is there any (future) feature list, because realtime transfering is
ofcourse cool, but some basic management like 'save this phone call',
CRM etc. would by nifty so you can use it for callcenter purposes too.

> If your interested please let me know. Im gonna be putting up a site for
> downloading if there is enough interest.
Very interested... and new around here ;)

> We are considering writing a SIP client build into the program at a
> later time.
Maybe IAX is a better choice is it is an * specific product?



Greetings,

Stefan de Konink

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[Asterisk-Users] No ringing sound on GS phones

[Stefan de Konink]

Hi,

The same problems occurs at our Red Hat system after the upgrade from
0.7.2 to 0.9.0. I didn't tryed the Grandstream phones, but our SIP enabled
Cisco 79xx's. Though a other funny thing happened what probably has the
same origin.
A Leadtec 'SIPPhone.com' box, had the opposite problem, while makeing
outgoing calls it has two ringing sounds at 0.7.2.

The only ringing sound we get at the moment, for all our phones is while
making outgoing calls. (So all local calls are mute) I'm going to try the
'r' add after the standard extention to fix this. But a permanent
solution would be nice :)


Greetings,

Stefan de Konink
The Netherlands

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