I am trying to compile * on SuSE 8.2. When doing the make install in
/usr/src/zaptel I get the following error.
**
/usr/src/linux/include/asm/system.h:189: warning: dereferencing
type-punned pointer will break strict-aliasing rules
freeIn
Hello All,
Thanks to those that responded to my problem of compiling on SUSE 8.2. I
was not able to get the compile done so decided to put RedHat 9 on this
system. After getting a RedHat supported NIC and RedHat installed,
Asterisk compiled cleanly, one SIP phone is connected and voice mail
I am getting the following messages that seem to be coming from Asterisk.
In the system there are no ZAPTEL cards installed. I did uncomment ztdummy
in the Makefile in /usr/src/zaptel before running make install. Any
ideas on how to get rid of this message. I looked through all the config
files
My config that works for number 1 is below. Everything works including
the voice mail waiting light. All of this for * was copied from or based
on:
http://www.automated.it/guidetoasterisk.htm. This is an EXCELLENT getting
started site. Can't help you with #2 but am sure others can.
sip.conf
I was incorrect in my citation of credit in the below email. Properly the
credit goes to John Todd for the Asterisk config examples. His excellent
article is at:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1
Sorry for the goof-up.
Robert
My config that works for number 1
Do you have a 100 or 101? You have indicated different models in your
postings. Were you able to get Call Transfer and Call Waiting working
with your Asterisk system and other phones? Which version of the
Grandstream firmware do you use? There most recent on their website this
weekend was at
I only have 1 but the absolutly only time it has to be rebooted is when I
change a parameter or upgrade the firmware. It has run for weeks without
any problem. Another poster mentioned the 10 vs. 100 Ethernet speed.
Maybe Grandstream can upgrade the interface in future hardware. I don't
imagine
Michael,
That would work for me too. If the volume can be reduced (maybe to zero or
almost zero) then my request for the ability to disable it is not needed.
Since the volume of the speaker and handset can be controlled maybe the GS
folks can include a patch in the next release of the firmware
The only thing that is wrong is that there seems to be some expectation of
Digium that they have to tell things... The source code is available. If
someone isn't happy with the Digium methods then they should find a
solution and post it to the list and/or one of the several Asterisk Wiki's
that
Andrew,
I am running it rather well on a original Pentium 100 Mhz, 32 MB RAM, no
USB adapter. I agree with you this would not be an ideal setup for a
business but in a home it will work rather well. I think it'll handle 2 CO
analog lines fine.
Yes, my wife thinks its overkill. Probably is, but
Chris,
Good point. As I understand it, the Asterisk software requirement was to
be a PBX between normal telephone lines and VoIP. Maybe even it was just
to replace the expensive PBXs. As such seems to me that it clearly met
and exceeded its design requirements since it utilizes the hardware
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference room the following message is played:
That is not a valid
On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference
Did you modprobe ztdummy before running asterisk ? I have meetme
running in one * box without zaptel harware.
I just tried that.
The following messages are given:
/lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such
device
Hint: insmod errors can be caused by
Asterisk...
Linux...
You get what you pay for. And it's free
:P
Thats true but free (cost) doesn't have to mean cheap (quality). Maybe
what we need is to collect business requirements and build a configuration
for a typical system. (hardware spec. and actual config files) What Dave
has
look at the rtc driver then. you do have a rtc chip already on the
system.
I looked back in the list and looks like the message that mentioned who
wrote ztrtc I deleted. Can someone please let me know where to obtain
ztrtc? I did a google on it and came up empty.
Thanks,
Robert
Seems you used my abreviation. It is really known by zaptelrtc. It seems
to be written by Klaus-Peter Junghanns [EMAIL PROTECTED] and is
distributed at http://www.junghanns.net/asterisk/.
Thanks for the info Steve. I got it but the make didn't work. Will work
on it over the weekend.
Not
I have my sip phones going into the context [from-sip] and would like to
play an introduction message and then have the caller enter the extension.
The message (dir-info was picked just to have something) doesn't play.
Maybe I misunderstood the s extension. According to what I read it is
The s extension is used when there is no known called number. In
other words, if you are dialing 2000, the dialplan will always prefer
the priority list for 2000 instead of going to 's', so that is why
your current system doesn't work.
John,
Thanks for the details. Actually what I want
7 - Ringer volume control
4 - plug in module of user programmable buttons for frequently called
numbers. Not everyone would need this so being able to add as an
optional module would keep the base phone cost effective.
9 - ability to switch back and forth between speakerphone and handset
7
On Tue, 21 Oct 2003, Low, Adam wrote:
Maybe I am missing something here but why would it downgrade their
network speed to 10mbps, its very rare to find a 100bT switches these
days that don't also support 10bT. In a switched ethernet network there
would be no performance loss for the other
Are you manually updating the mySQL tables or do you have a web app. to do
that?
Robert
Steve Creel wrote:
You'll want to #include it. This leaves the burden of the [general] and
any static configs on sip.conf but allows the script to blindly write out
from the database to sip_additional.conf
Ouch.. you hit one of my pet peeves.. See below.. This is not meant to be
a FLAME but rather SOAPBOX.
Robert
John Brown (CV) wrote:
http == hyper text transport protocol
So are the entries on your hard drive with a .htm or .html extension not
files? (sorry a little sarcastic I know)
***
Is anyone else having trouble receiving IAXTel calls? I don't know if
it's my config that's broken or IAXTel that broken. Several people have
given me their IAXTel numbers and calls to them all fail. I can call
FWD numbers via IAXTel just fine.
--Eric
Eric,
I am having a similar
...
Still: When I call my Asterisk box (which has a fixed IP and is located
within a university network) using X-Lite I get choppy sound to say the
least. In fact I can hear only the first half second of what I am
supposed to hear followed by permanent silence. Note that this * box has
no
Can anyone please point me toward the source/binary (linux and Win32) for
Gastman??
Robert
Hi,
The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all
my machines, no error, no log.
Although, the CVS version works great on Linux.
Is it anybody who knows how to compile it
Jarad,
I would be interested in one or 2 of your examples to get an idea of how
to get started.
Thanks,
Robert
Friedrichshafen, Germany
On Fri, 2003-10-24 at 05:54, WipeOut wrote:
First off, can AGI scripts be created using PHP??.. This is where our
skills are and since PHP can be run from a
Okay, at the CLi i did a show version and it's still showing the old
version. What I'm attempting to prevent the overwriting of my already
established config files and sound files. Any further suggestions?
When I did the make on Asterisk the first (and only) time, I had to do
make samples
...
At this moment, Asterisk behind a NAT can't connect to an outside SIP
provider. If you put asterisk outside your NAT, your inside clients
can connect to Asterisk and Asterisk will be able to connect to your
providers.
I suspected this would be the case. The problem is that I have no
Alexander,
I will be happy to help with the testing but since I am behind NAT am not
sure it will be of much help to you.. I have 2 Grandstream phones and
Asterisk.
Robert
Friedrichshafen, Germany
Hello All,
We are looking to test interoperability between Asterisk and Nextone
softswitch.
My asterisk server(s) are behind NAT, and I am a customer of Vonage
(thrice-over), iconnecthere, and Net2Phone.
There are still some rough edges (especially with iconnecthere) but
overall it is not correct to say that they won't work.
B.
Thats great to hear. Can you please share your
I have an Asterisk box behind NAT and am trying to connect to Iconnecthere
as was indicated possible earlier. Am getting the following on the
Asterisk console:
-- Executing Dial(SIP/2001-12c8, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
== No one is available to
Interesting. Someone thinks that a strategic use for * should be off
this list. Someone thought my FAX modem for * should be off this list.
However, nobody seems to think a 1000 messages about Grandstream phones
should be off this list.
Personally I would welcome seeing more of what people
])
This works for me
regards
Miklos
- Original Message -
From: rnc Info Lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, October 25, 2003 5:17 PM
Subject: [Asterisk-Users] Iconnecthere connect problem
I have an Asterisk box behind NAT and am trying to connect
I have a SwissVoice IP10S but can not seem to get it to have dialtone or
dial on *. Calls to or from 3001 don't work.
Any ideas are appreciated.
Robert
mgcp.conf is:
[general]
port = 2427
bindaddr = 192.168.0.110
[ip10]
host = 192.168.0.5
context = from-sip
line = aaln/1
The portion of
Citeren rnc Info Lists [EMAIL PROTECTED]:
I have a SwissVoice IP10S but can not seem to get it to have dialtone or
dial on *. Calls to or from 3001 don't work.
Were you able to configure the phones through their webinterface ?
You could try entering 'mgcp debug' and then power up your
Hi,
-Original Message-
The portion of extensions.conf is:
exten = 3001,1,Dial(MGCP/aaln1,20)
exten = 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20)
Or aaln/1@ip should do just fine. However this doesn't explain why there
is no dialtone on the phone..
Oh, one thought: Did you set
As far as I know they do only SIP. If your Asterisk box is behind a NAT
firewall then you probably will have problems.
Hi All,
I have a FWD number and wish to connect it to Asterisk to receive my FWD
calls.
How I do?
Is it a register in sip.conf or iax.conf?
Regards
Dave
html
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.
Hi ,
I even think to avoid using an installer mainly because the installer
part is bigger that the application himself.
What do you think?
Dan,
I agree that if an installer or registry entries are not needed then it
makes an automated rollout much easier. Also makes it possible to run
On Mon, 2003-11-03 at 16:27, Alastair Maw wrote:
On 03/11/03 20:03, Steven Critchfield wrote:
Sounds like you really need a C programmer and get into the guts
of asterisk. Can't get more flexible than having the source code
yourself to do anything you want. You could add your DSP routines
Daniel,
the MGCP log you sent shows you sending the digits and asterisk receiving
them, however after that either nothing happens (infinite digittimeout) or
you cut the log short. Can you also send some console output with 'mgcp no
debug' :-) It saves clutter. Maybe a peek at your
With some lively IP10S discussions here maybe someone knows about this
issue: I can use the speaker phone ok. However the handset and switch
hook do not seem to work. If I enable headset then I can get audio via
the handset but still have to use the speaker phone button to take ot off
hook.
Hi Robert,
I haven't the HeadSet model but the lan switch model so I can't be of
any help for you.
Daniel
I have the IP10S LAN Switch model too.. Thats why I find it wierd that the
headset setting makes the difference !
Robert
___
Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net
I've posted my demp weather report Asterisk AGI script at
http://www.fnords.org/~eric/asterisk/downloads/
Eric,
Can you comment on the difference in installation ease for Festival and
Cepstral?
Regards,
Robert
Cepestral was installed and working within 10 mins of my decision to
purchase it. It's $30.00 and can be purchased on their web site and
they give you a download. They have a demp on their website that will
do text-to-speech and give you a .wav file to download and listen to.
Download,
Earlier today someone posted a RFC number related to voice mail.
Unfortunatly I deleted the message so have lost the number and don't see
it yet in Google. Can you please resend that to me?
Thanks,
Robert
___
Asterisk-Users mailing list
[EMAIL
I think it is time to start commercial Pro version (not expensive !!!) of
Asterisk.
In my company we already made decision to do it, to offer people
ready-to-go solution. But is is hard to do anykind of such product without
Digium and Mark's support.
Mark I think you are very overloaded
Is anyone running Asterisk under Fedora Core 1
(http://fedora.redhat.com/)?
If so, did everything with Asterisk work properly? I'm looking to migrate
from Red Hat 8.0 to Fedora this week.
Thanks.
Interesting question... Since RedHat will in the future have only their
Enterprise version I
Max, That is what worked for me. if you want the MESSAGE button on the GS
to dial the VM then put whatever extension you have defined for VM in the
field Voice Mail UserID via the GS Admin Web Interface.
Robert
Hi Folks,
Bit of a newbee here, so please be gentle. :)
I'm trying to get
Does anyone know of SIP phone providers (Grandstream in particular) who
are located in Germany (or the EU)
Thanks for any info.
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I have a problem where the Background application only seems to work if
one digit is pressed. Extensions with multiple digits just timeout and
asterisk hangs up.
Below is the relevant excerpt from extensions.conf. In this example,
pressing 2 will access the service menu. Then pressing 1 will
Why don't we just add it on the DIgium list server, wouldn't that make
more sense, to have a single place for all list memberships?
Mark
OR even just leave the discussion on asterisk-users... If we create new
lists everytime some people disagree with a topic being on-list then we
will have
Are you also able to make outgoing calls via Iconnecthere? If so do you
mind posting your config? I tried their 10 minute trial a couple of
months ago but was not able to get a connection.
Thanks,
Robert
I'm receiving calls on my asterisk server from iconnecthere. My asterisk
server is
Does anyone have experience using the Netphone SIP phone from Ortena
Networks (http://www.ortena.com). I contacted them, and they will sell
me 10 units for 95 euros/unit. At least i -looks- better then the
Grandstream :-)
The phone looks interested and appears to have been on the market for
Hey, surprise! Just discovered it on the web:
http://graphics.cs.uni-sb.de/~rainer/tour.jpg
Mark is going on tour!
Not sure if this is real info or just a JPG that someone created.
Is Stuttgart a definate date on the 30th? If so, where in Stuttgart??
Robert
Friedrichshafen
Is anyone other than me having trouble dialing out via IAXTEL? I havn't
changed my config files in weeks but seems that IAXTel calls (800 and FWD)
stopped working in the past week sometime.
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Yes, I've been having problems as well but had not taken the time to
diagnose
the problem. Just did some looking and it appears iaxtel.com has removed
the iax v1 support. iax2 seems to be working fine.
Rich,
That solved the outbound problem.. Thanks for the hint... 800 numbers are
accessable
On Fri, Dec 12, 2003 at 01:57:02AM -0500, Brian Capouch wrote:
John Brown (CV) wrote:
Hi List,
Just a quick note that we have cleared all back logs of Grandstream
product. If you have been awaiting shipment, its shipped. Everyone
should be getting tracking numbers shortly.
We
it's a firmware problem on GS, they are working on that but it seems its
not that simple to make volume higher on the speaker and echo go away,
anyway 4.26 seems stable for now and with many new features!
Miguel,
What are the new Features?
Robert
___
I tried again at runlevel 3 but to no avail.
I'm pretty sure I have sufficient horsepower since I'm running on a box
with
half gig memory and a speedy CPU.
burak
I run on a Pentium I /100 Mhz, 32MB RAM with RedHat 9.0 and have no
trouble with voicemail audio or Music On Hold. This is a
Today I deleted the files in the asterisk, libpri, zaptel directories that
are in /usr/src and did a new CVS checkout (not update). After doing
the make installs and starting asterisk the show version is the same
as before:
Asterisk CVS-10/09/03-20:33:57 built by [EMAIL PROTECTED] on a i586
On Sat, 2003-12-13 at 16:41, Joe Dennick wrote:
I just updated yesterday, but I did a complete rm -Rf for all of the
following directories:
/usr/src/zaptel
/usr/src/zapata
/usr/src/libpri
/usr/src/asterisk
Then I did a new cvs checkout for all four of those items
Hi!
I don't get why people always say dtmfmode=info mine works fine with
rfc2833.
bkw
Dunno. I tried rfc2833 first, and had exactly the same problem as
described below with voicemail (but only there). Info then worked just
fine (as obviously also confirmed by this user here).
Is there
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream Quality Survey :P
Reply-To: [EMAIL PROTECTED]
...
I have 2 of these phones and they work fine for my application. Granted
its not the most intensive use and definatly not the most critical users
Message: 11
From: Asterisk online forums [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P
Date: Wed, 24 Dec 2003 11:23:14 -0500
Reply-To: [EMAIL PROTECTED]
Brian,
...
We are looking now to improve GS products and start collecting
The phone powers up and I can make calls through my Asterisk gateway to
other endpoints. However the four leds under the keypad are permanently
illuminated and the backlight slowly flashes on and off. When I pick up
the handset there is a repeated tone before I get a dial tone.
I know it's
Still, there seems to be a you get what you pay for theme to many of
today's posts and this clearly applies to support on FWD. Naybe we should
remove the signature from * that enables FWD to identify * systems :-)
That certainly seems the case for today's theme... It is certainly the
right
Hi there,
yesterday I came across the Vocera Communication Badge and now I'd like
to know if anyone here has played with that thing (or even just seen it
in real life), and if a price tag can be found for this device?
Too bad they don't use SIP... ;-(
http://www.vocera.com/
I'm trying to buy a new X100P but
http://shop.store.yahoo.com/bsdmall/wisifxoin.html
is failing to check the order
Anybody knows any other way to purchase it?
Isamar
Try http://store.yahoo.com/asteriskpbx/wildcardx100p.html
You won't get the whopping 95 cent discount from BSD Mall but
Where can I find that Howto? I'm new to Asterisk and am looking for all
the
doc I can find.
TIA,
Eric
Eric,
You will find at at:
http://members.lycos.co.uk/wipe_out/asterisk/
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Check http://www.telappliant.com for their VoIP Starter kits or Telephony
Cards sections.
Robert
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello there,
.
for pointing me at a friendly/knowledgeable UK supplier of such cards.
Any advice would be greatly appreciated: once I have
Sorry 'bout that.
-Original Message-
From: Kris Edwards [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 06, 2004 3:38 AM
To: '[EMAIL PROTECTED]'
Subject: Matrix Orbital (usbl LCD or VFD)
This probably isn't practical for anyone other than home users, but I
would like to use a
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