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
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:unoffic
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
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.
W
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 ce
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
e 7940/60 or does this still
require renaming the file to .cgi? (With handler support)
Greetings,
Stefan de Konink
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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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-DB/FromPeer)
Greetings,
Stefan de Konink
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ertification 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
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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=95
http://www.scansoft.com/realspeak/demo/
Stefan de Konink
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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?
Stef
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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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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[
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:
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 in
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 whe
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
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, an
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
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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ull 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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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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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 cod
729, 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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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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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
/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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To
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 K
r 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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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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f 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 K
also includes the phones firmware sources?
Stefan de Konink
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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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ave 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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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
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
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
C 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
__
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, EnterPas
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
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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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:
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 vi
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
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 si
u, 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: Thurs
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)
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] LDA
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
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 feature
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
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 file
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)
Ther
e (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
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 Hoyl
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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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
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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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 la
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 tha
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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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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a permanent
solution would be nice :)
Greetings,
Stefan de Konink
The Netherlands
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