At 15:07 1-11-2003 -0600, you wrote:
Last I checked skinny firmware would try to connect to a host that would
resolve to CiscoCM1
Actually that is just a last-resort. Before that it will try and find the
callmanager by looking for some special DHCP flag, and if that is not
around it will try
WipeOut wrote:
David Sussman wrote:
Apologies if there is a cleanly written and searchable FAQ that I
could be
directed to. I have no problem to RTFM if I can find the FM...
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under
Rich, thank you for your informative reply. I checked with our admin and he
replied:
I setup from the start nat=yes and canreinvite=no on sip phones from
Internet and modified the rtp channels (voice ports) and the rtp
port on the phones. Still have the same problem, no sound.
Perhaps the VPN
Hi-
I'm looking for an appropriate system board to power a system with two (2)
Digium TE410P cards. Since these cards require the 3.3 volt PCI, I'm
considering vendors like Tyan for the motherboard.
Can anyone please tell me their experiences with the Tyan i7501 series
(Xeon-basd), or recommend
How
do you suggest doing that?
How
can I convert wav files to gsm files?
thanks
Shoval Tomer, MCSE
IT Manager
Softov Advanced System Ltd.
Email: [EMAIL PROTECTED]
Mobile:
972-55-229220
Shoval Tomer wrote:
How do you suggest doing that?
How can I convert wav files to gsm files?
thanks
#!/bin/sh
for i in *.wav; do sox $i -r 8000 `basename $i *.wav`.gsm resample -ql; done
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi Scott,
I use a Tyan 2665 (7505 based) M/B with a TE410P. That works well. This
is a development workstation, so its probably not the kind of board you
want for deployment.
Regards,
Steve
Scott Stingel wrote:
Hi-
I'm looking for an appropriate system board to power a system with two (2)
Has anyone tried running Asterisk under Trustix? (or Tawie as it is now
called)
Later..
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Either it's not working, or I don't know what I'm doing. It's giving me the
error sox: effect '.gsm' is no known!
Let's say I need to convert file 1.wav to 1.gsm.
How do I apply this command to it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel
Shoval Tom wrote:
Either it's not working, or I don't know what I'm doing. It's giving me the
error sox: effect '.gsm' is no known!
Sounds like your copy of sox was not compiled with gsm enabled.. or you
put a space between the ...wav`.gsm bit
check with a single file like this:
$ sox
Hi all,
Still having the one way sound problem.
Any suggestions how to hunt the problem down ?
Regards,
Thorsten
---
Hi all,
We have a very basic * installation for testing purposes.
The * is connected to PSTN with BRI and setup with
Hi Steve-
Yes, I was looking more for a less robust board (with integrated AGP) that
would be more appropriate in a 2U rackmount for my customers - don't need
firewire, SCSI, USB etc). I didn't really want to go to the Xeon at all,
except that it seems that the 3.3v PCI requirement seems to push
Netfinity 4000R
servers that do not support X windows under RH8.x and I
prefer not to go
back to RH7.3...
I recall in the archives somewhere, and through someone's
post earlier
today, that there is some sort of problem with RH9 with
Zaptel (hardware)
drivers and that RH8 is
Netfinity 4000R
servers that do not support X windows under RH8.x and I
prefer not to go
back to RH7.3...
I recall in the archives somewhere, and through someone's post earlier
today, that there is some sort of problem with RH9 with Zaptel (hardware)
drivers and that RH8 is
All of the setup is running on RedHat 8.0 - no other router
or CSU is needed.
Don't use RedHat 9.0 yet in this setup since the
ZAPTEL_NETWORK flag will not compile with the new
implementation of HDLC in the kernel.
I believe that when you use up2date on both RH8 and RH9, you end
I can also confirm chan_h323 and g.729 work well to 5300s, but we had
to build on RH8 not RH9. Haven't tried 5300 to Asterisk
except via SIP
which works fine--even to i4l interfaces.
I believe that when you use up2date on both RH8 and RH9, you end up with the
same version of Kernel. So
unsubscribe
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To try and put it simply, the zaptel drivers will not compile with the
-DZAPTEL_NETWORK flag (as set, not my default, in the Makefile), with any
stock kernel including and after 2.4.21, which is when the new HDLC structure
was imported from the development kernel tree.
Therefore, it should be
Shoval Tom wrote:
Either it's not working, or I don't know what I'm doing. It's giving me the
error sox: effect '.gsm' is no known!
Let's say I need to convert file 1.wav to 1.gsm.
How do I apply this command to it?
FAQ. See
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
(I've just
Hi!
reinvite=no
canreinvite=no
Don' these options have the same meaning? Just wondering...
P.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I think I found another interesting 'feature' with threeway calling. If you
hang up while on a 3 way call with both parties on outside lines, Asterisk
ends up removing the conference initiator and leaving the outside trunks
bridged together. Is this a good idea? This could cause congestion
So true, yet so irrelevant for my purposes.
I needed to convert existing IVR sound files to gsm, in order to demonstrate
asterisk's functionality to my bosses (the ones who'll pay for the hardware,
eventually...)
Besides, even if I didn't have the files ready, I wouldn't use my lovely
voice for
Olle, I can't reach the faq page, and haven't been able to for the last four
days.
I'm getting 504 gateway timeout errors.
Any ideas?
Btw, the first answer I got worked, I mistook ` for ' (newbie error, I
know...)
To be more specific for you newbies out there
Create a file containing:
Hi all,
I have developed a full featured Windows IAX phone based on LIBIAX library .
It is now in a prerelease version (0.9.0) and you can download it for free
from my web page:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
Some of the features are:
- registering with Asterisk
Besides, even if I didn't have the files ready, I wouldn't use my lovely
voice for it - I'll go to a recording studio with a professional (talking
about a production environment) so it's good to know how to do this
yourself, in case the studio doesn't know how to record them in this
format.
Sorry... I was a bit in a hurry, and indeed I cannot expect all list
readers to know about shell scripts... will elaborate a bit more in the
future.
I noticed you removed the sox resample -ql options, which on my studio
recorded .wav files helped a bit, also It might be sensible to add a -c
1
Shoval Tom wrote:
Olle, I can't reach the faq page, and haven't been able to for the last four
days.
I'm getting 504 gateway timeout errors.
Gateway timeout indicates something with your web proxy ...or?
I've been able to reach the Wiki all weekend, I've updated and created
several pages...
I
Hi,
I
started looking into asterisk cause we're looking for a real-world solution.
(when
I say we I talk about a 50+ HQ and a 10+ branch office).
We currently
use a Panasonic analog PBX, with home-made IVR and PSTN lines.
We'd
like to deploy most of Asterisk's capabilities
Dan,
Looks great.
Are you planning to release this with GPL?
Peter
At 22:21 2/11/03 +0200, you wrote:
Hi all,
I have developed a full featured Windows IAX phone based on LIBIAX library .
It is now in a prerelease version (0.9.0) and you can download it for free
from my web page:
How should I configure Asterisk to allow this soft-phone to register?
Please provide an example
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of rnc Info Lists
Sent: Sunday, November 02, 2003 11:23 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Hi,
I believe the issues raised by this message are the same as mine, more on a
commercialsense than for self use, but mostly the same. I've seen posts
where real-life installations are mentioned, but not a reference to how Asterisk
is working on production (and productive) environments.
Hi,
Here is wath happens:
Asterisk*CLIsip debug
SIP Debugging Enabled
Asterisk*CLI
Nothing happens when I use 'sip debug'.
It seems that sip doesn't work.
Wim
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 9:02
I recall in the archives somewhere, and through someone's post earlier
today, that there is some sort of problem with RH9 with Zaptel (hardware)
drivers and that RH8 is preferred.
Do you recall what kind of problem? The only problem I have is an annoying
echo that I haven't yet gotten rid
Anyone have any example scripts in PHP that connect to the manager? I'm not really a
much of a programmer so I could use boost. Once I can figure out how to get it to
login properly, I'll be ok from there.
Thanks,
Kevin
_
Are you a
As the library is under LGPL (is not true?), I intend to keep this
application as a freeware only...
Yep its LGPL.
Play with it and try to use all the features, which are very intuitive.
Its a start but having to restart when you change registration isn't very
intuitive. But its an
Hello Steve,
You are exactly right about the DDI line and thank you for clearing up
the 600 ohm loop.
Can you tell me other electrical details? It's just that all the PBXes
(that I know) uses
different cards for DDI lines and analog extension lines and since the
CO normally or at
least,
Finaly, someone has started the IAX soft phone ball :)
Thanks, Dan...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Kevin Bockman wrote:
Anyone have any example scripts in PHP that connect to the manager?
I started a PHP * Manager API, modeled on the Perl API, but haven't had
a lot of time to work on it. I'll be happy to give you what I do have.
--
JustThe.net Internet New Media Services
22674 Motnocab Road
Is there a way to clear the Queue stats?
That is with out restarting *?
I'd like to reset them daily and don't see a way
to do that.
Unless the only way is maybe a cron restart asterisk
like every weekday @ 04:00?
___
Asterisk-Users mailing list
[EMAIL
--- Ken Godee [EMAIL PROTECTED] wrote:
Is there a way to clear the Queue stats?
That is with out restarting *?
I'd like to reset them daily and don't see a way
to do that.
Unless the only way is maybe a cron restart asterisk
like every weekday @ 04:00?
Finaly, someone has started the IAX soft phone ball :)
Thanks, Dan...
actually theres been an opensource multiplatform iax soft phone on sourceforge
for a while now:
http://iaxclient.sourceforge.net/
duncan
___
Asterisk-Users mailing list
[EMAIL
--- Steve Sobol [EMAIL PROTECTED] wrote:
I started a PHP * Manager API, modeled on the Perl API, but haven't had
a lot of time to work on it. I'll be happy to give you what I do have.
___
Sure, I'd appreciate that. All I really need to start is to
It would be nice to see a real extensions.conf
from a live business operation, every extensions.conf I've seen posted
or been able to dig up so far would fail bad in a live business operation.
I just have the beginings of mine and would like to make sure I don't
miss anything.
Most
Hi,
I started looking into asterisk cause we're looking for a real world solution.
(when I say we I talk about a 50+ HQ and a 10+ branch office).
We currently use a Panasonic analog PBX, with home-made IVR and PSTN lines.
We'd like to deploy most of Asterisk's capabilities through out our
Thanks for the detailed answer, and sorry about the not so detailed
question.
So here's my humble request.
Can someone who has implemented a live production Asterisk deployment,
preferably between two sites (HQ and a branch office, connected over the
internet) spare the time and contact me here,
WipeOut wrote:
Jason A. Pattie wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ariel Batista wrote:
| Ok I have a question. I have Xten-lite working with our Asterisk
system and I am able to make and get calls. But the main problem is the
sound is very choppy and sometimes it cuts off
So here's my humble request.
Can someone who has implemented a live production Asterisk deployment,
preferably between two sites (HQ and a branch office, connected over the
internet) spare the time and contact me here, or to my email directly?
As a lurker, I would very much appreciate if this
Interesting thought, with these DDI lines a UK based company could easily
get a good number of incoming analogue lines into an Asterisk system
because teh FXS cards have far more ports than the FXO ones.
Michael
___
Asterisk-Users mailing list
[EMAIL
i've already tried to change the indications.conf to the following:
dial = 0/1500
but the dial tone still persists
i am using the following workaround but obviously not a clean
b'cos it just replace dial tone with some other tone.
in zapata.conf
context=spec
immediate=yes
Problem I have is this. outside firewall
(extension 2003) can call me inside firewall (extension 2000) and all is
fine. If I call from inside firewall (extension 2000) to outside firewall
(extension 2003) I hear no ringing and person at other end can pick up and I
hear for maybe a half
Any thoughts or plans on making it available on the asterisk key *NIX?
AJ
On Sun, 2 Nov 2003, Senad Jordanovic wrote:
Finaly, someone has started the IAX soft phone ball :)
Thanks, Dan...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
As the library is under LGPL (is not true?), I intend to keep this
application as a freeware only...
I want to add new features, but for one of them I need new functions
implemented in the library (like multiple codecs support, message waiting
indicator, conferencing, etc.).
There is no
(The list may get this msg twice; I originally sent it from
the wrong email address, my apologies. Moderator, if you can,
please delete my original email submission from
[EMAIL PROTECTED] Thanks.)
Hello-
Summary:
Can anyone recommend a place to start to learn how to create an Asterisk
system
Hi Patrick,
From memory (I haven't lived in the UK for 11 years) the electrical
characteristics are pretty much the same: -48V, 35mA loop current,
600ohm complex impedance. One key difference is an extension needs to
ring, but a DDI line does not. The different cards you see used may be
hello,
I am completely new to things but was wondering if some one could steer
me in the right direction [i.e. i was volunteered to get a PBX running
with little or knowledge] good news is, i got a lot of experience with
open source / linux / etc. anyhow. we have 4 lines coming in and need 16
Hello,
I am posting this after spending hours digging through the list archives.
Problem : When asteirsk plays a voice prompt, the voice clip is really
choppy.
I figure that this is something to with the sound card, the timing of
playback etc.
But cannot seems to find an answer.
Here is the
Here is my example. I'm using a lot of times a day.
?php
$socket = fsockopen(192.168.0.53,5038, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: blabla\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket,
On Sat, Nov 01, 2003 at 09:35:26AM +0100, Florian Overkamp wrote:
At 01:43 1-11-2003 +0300, you wrote:
Is msn messenger capable of using asterisk as it's gateway?
Yes, provided you are using MSN 4.7, and not 5.0 or higher. Configure the
Communications Service under the Options/Accounts
I built something very similar using:
- Adtran TA750 bought off Ebay for around $400 (you can do much better,
I was in a hurry.)
- A Digium Wildcard T100P
- A 4 port FXO card for the TA750 (I searched Google for Adtran FXO
and clicked one of the sposored links.)
You might have to pick up
Hi again,
thanks a million for the info.
regards,
patrick
Steve Underwood wrote:
Hi Patrick,
From memory (I haven't lived in the UK for 11 years) the electrical
characteristics are pretty much the same: -48V, 35mA loop current,
600ohm complex impedance. One key difference is an extension
On Sun, Nov 02, 2003 at 05:56:02PM -0700, Matthew England wrote:
(The list may get this msg twice; I originally sent it from the wrong email
address, my apologies. Moderator, if you can, please delete my original email
submission from [EMAIL PROTECTED] Thanks.)
Hello-
Summary:
Can
Hello
all,
Been a while since
I've strolled this way.Apologies in advance if this is a common line of
questioning.
I've just bought a
new Intel 865G based board with a P4 Hyperthreading
processor.
I believe I've
gottenSMP set up correctly: in the menuconfig I specified SMP and told
Hi Dan.
thanks for good application!
and I wish 'no with installer' package about that.
because I think use with USB-memory device in any places (ie.net-cafe.)
is that need registry setting or not?
On Sun, 2 Nov 2003 22:21:09 +0200
Dan [EMAIL PROTECTED] wrote:
Hi all,
I have developed a
Robert,
Try adding canreinvite=no to extn 2000 and reload asterisk. In your
specific case, it needs to be on each sip.conf extn definition.
Rich
Problem I have is this. outside firewall (extension 2003) can call me inside
firewall (extension 2000) and all is fine.
Hello,
I would like to setup an * system but have no
experience with Linux and am just learning about VoIP. My programming
experience is pretty limited as well, so I may be getting in way over my head,
but I am willing to take the time to figure out how to use *.
I'd like to use * to
Hi Mark,
There is no requirement that you GPL or LGPL your code (other than the
requirements that you publish changes to iax-client and/or libiax.
There is no change in the libiax for the moment. My DLL is just used to
export the functions from the library to the main application.
However,
66 matches
Mail list logo
