Hi List,
I have managed to compile asterisk but I can't start it. What I have
done so far as asterisk config is concerned is cut and paste the sample
config files from the ONLamp article on Asterisk.
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
When I start asterisk -vvvp I get
Hello ,
Has anyone explored video conferencing on Asterisk with SIP ? I dont
want to use H.323 as everything else is SIP based in the set up.
I have gone through the lists but there doesnt seem to be any info
about video conferencing with SIP.
I dont have any users dialing in , but every one
I was wondering how the reload option in asterisk (asterisk -r -x reload)
affects calls in session and other activity like active AGI aplications. I
tried it using a single call which i placed to my asterisk box and it didnt
get disconnected when i reloaded asterisk.
But what about heavy load
So what? You said you had ssh access. Use ssh forward.
Here's another way. Might work for you, I don't know.
You could easily setup a secure tunnel (think openvpn) to run your ftp
server on locally. That way you could keep all the configs in one place.
You could open the tunnel when it's
On Sun, 31 Oct 2004, Remco Barende wrote:
I will probably order the base station, it seems like an almost ideal
solution to connect phones to a voip pabx. I would not prefer a pci card
solution personally, anything connected to the network doesn't cause irq
headaches :)
On the other hand
I think that the topic is sidetracked...
My original question is about how to change the default username and
password for ftp login. I want to change it, but don't want to punch the
keypad manually. I don't think that this can be done via web interface
either.
Richard
-Original
On Sat, 30 Oct 2004 21:15:51 -0400, Steve Kann [EMAIL PROTECTED] wrote:
The chart is good, but I think it makes a mistake for iLBC:
Isn't iLBC 13.something kbps?
Also, since iLBC uses 30ms frames (when used with asterisk, at least),
it has slightly lower overhead. Approx 2/3 as much
On Sun, 31 Oct 2004 08:18:22 +0100, Vikram Rangnekar
[EMAIL PROTECTED] wrote:
Also does
frequent reloads affect the stability of asterisk i mean things does it lead
to things like memory leaks
Depends on the version of Asterisk you are using and your environment.
I have seen frequent reloads
Hi,
-Original Message-
Has anyone explored video conferencing on Asterisk with SIP ?
I dont want to use H.323 as everything else is SIP based in
the set up.
I have gone through the lists but there doesnt seem to be any
info about video conferencing with SIP.
I dont have any
Hi All,
I was doing some testing between on extension running SIP at G.711alaw and
an IAX extension runing iLBC (also GSM) and found that the voice from the
IAX user has a lot of packet loss (very bad voice quality) toward the SIP
phone only. From SIP phone to IAX, voice quality is fine. SIP phone
I've written a small AGI thing to allow lots of stuff, including
diverts. If a call is placed to a diverted number, a new call is
initiated from * to that number. Simple. But then, to make billing
sane, I need to change the 'dst' in CDR to reflect the number
diverted
to.
How can I do this?
I
On Sun, 31 Oct 2004, Garry Taylor wrote:
Hi All,
I was doing some testing between on extension running SIP at G.711alaw and
an IAX extension runing iLBC (also GSM) and found that the voice from the
IAX user has a lot of packet loss (very bad voice quality) toward the SIP
phone only. From
Hi,
I'm testing different VOIP hardware with asterisk and try to transfert and
hold a call.
My test with SIPphone (grandstream BT and cisco 7940) and softphone
(sjphone) are ok when I'm using dtmfmode=info.
But with FXS devices (GS Handytone and Vega50 FXS) and very simple phone
(10 digits, #, *
IAX extension, ie. firefly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, 31 October 2004 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] G.711alaw to iLBC
hi
we're interested in CallerPres in chan_sip.
what will it take to implement it?
roy
On Oct 25, 2004, at 4:37 PM, Race Vanderdecken wrote:
Roy et All,
If someone could expand on CallerPres requirements in chan_sip I
can do the work. I have added numerous extras to chan_sip already,
On Fri, 2004-10-29 at 13:41, Maxim Litnitsky wrote:
Hi all, I have to implement the following:
--
| 10 voice channels
|---|
Prov E1 | 256 kbit/s for VoIP |
Asterisk IP-PBX
Hi Mr Jack, hi everybody
Thank you for your answer for my message titled can't run ztcfg. I tried
what you proposed me and the error I told about is not signaled. However I
still have problems to get mfc/r2 support running.
I refered to the mfcr2 support documentation available in the
On Thu, 21 Oct 2004 09:39:48 +0900, Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED] wrote:
Wed, 20 Oct 2004 15:47:59 -0500, Henry Devito [EMAIL PROTECTED] wrote:
Where can I buy the act phones?
I have now discussed the matter of sample orders and shipments with
ACT directly and I have
All you really need is a list of 1-800 numbers in various countries.
Most multi-national corporations have a list buried somewhere on their
web site.
For example:
http://www.microsoft.com/resources/howtotell/ww/windows/what.aspx
Gilad ;-)
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A
My solution to this (as the debian package appears to actually download
mpg321 (instead of mpg123) when you install *, was to download mpg123
from the original website and compile/install it myself.
http://www.mpg123.de/
mpg123 0.59r is the version im now running (just copied the executable
in response to many queries asking for a URL ...
http://www.voip-info.org/tiki-index.php?page=ACT%20P104SLD%20IP%20Phone
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get
Thanks John, that worked. I guess that's a pretty common mistake :)
Now to build the rest of my config files, that's always fun.
On Fri, 29 Oct 2004 19:26:10 -0400, John Bittner [EMAIL PROTECTED] wrote:
I just read what I typed... I meant to say put the 614p in
the reg.1.address field with
Most phone manufacturers support Asterisk unless they also provide a PBX
product.
I have seen postings from snom employees on this list (they even sell their
own competing switch)
- Original Message -
From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED]
To: Asterisk Users
chan_capi.c:2603 load_module: Unable to load config capi.conf
You need to create this file /etc/asterisk/capi.conf
with the following content :
[general]
nationalprefix=0
internationalprefix=00
[interfaces]
msn=50
incomingmsn=*
controller=1
softdtmf=0
accountcode=
context=incoming
Just put a note that channels may vary do to placement of modules. I think
that would be more correct.
Also, try a different phone. I had this problem with a cheap cordless once.
Give us output from the console.
Give me SSH and I will have it working quickly.
- Original Message -
Dear Khaled,
I thing you must read the documentation a little bit more deapely!
does zaptel compile ok ?
which kernel are you using ?
have you configure the zaptel.conf file
which parameters are you using for r2 signaling ?
refer to this page as guide for starting
OK, Now I'm confused. It was working but I was using TFTP. I wanted
to use FTP so I just copied the config files to the ftp server,
changed the login info on the phone to FTP. Now the phone doesnt
login via ftp and get the config files but it won't even try to
register now. Has anyone ever
I'm prepared to kick off a bounty to get some form of video conference
meet me solution going.
My specifications would be for a minimum of 4 people in the conference
and to have some form of web page control, kick off-join-mute, mute all.
Is there some formal way of setting up a bounty on
Thanks for the tip! I'm still having a couple of quirks though...
Adjust devices= with the number of B channels supported by your card. For
ISDN BRI, it's 2, for PRI, it's 30.
Okay, I did that but then I had the exact error you describe below...
You need a kernel support for you card and you
Im
looking for a ISDN card that works under asterisk and supports BRI line.
And I just can`t findit. Momently im using card
INTERNAL, but Im having problems, asterisk on startup when loading modem
fails (i4l
driver).
Can you please help me, or
point to a www address where culd I find
Ok. Nevermind. For some reason this one phone won't connect to my
internal ip of 10.20.30.2 but it's able to connect to the external ip
where all of the other phones are able to connect to 10.20.30.2... So
that's an internal problem. So the configs do work.
On Sun, 31 Oct 2004 09:46:50 -0500,
You need a kernel support for you card and you also need to load a
firmware for some cards.
If you have a message like CAPI not installed!, check your kernel.
I use the linux 2.6.7 kernel which came with the knoppix distro I've
installed on the box. I have checked with make xconfig and
I get the following error when I try to compile asterisk on my redhat 9
box any ideas? CVS version from October 22, 2004
PIC -c -o pbx_dundi.o pbx_dundi.c
pbx_dundi.c:54:18: zlib.h: No such file or directory
pbx_dundi.c: In function `update_key':
pbx_dundi.c:1313: warning: implicit declaration
In a recent upgrade to version * 1.0.2 I have noticed
a new behavior in the Zapateller() function.
It now produces the 3 tones you get when you hear
the were sorry message from the phone company.
Anybody notice this New feature?
Regards
Greg Cirino
___
Cirelle
+++ Benjamin on Asterisk Mailing Lists [31/10/04 18:11 +0900]:
On Sun, 31 Oct 2004 08:18:22 +0100, Vikram Rangnekar
[EMAIL PROTECTED] wrote:
Also does
frequent reloads affect the stability of asterisk i mean things does it lead
to things like memory leaks
Depends on the version of
Does anyone have norwegian sounds (audiopack) for Asterisk?
Please send me an url for download if so. (sounds for voicemail too)
Best regards
LOH
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Cirelle Enterprises wrote:
In a recent upgrade to version * 1.0.2 I have noticed
a new behavior in the Zapateller() function.
It now produces the 3 tones you get when you hear
the were sorry message from the phone company.
Anybody notice this New feature?
SIT aka Special Information Tone is the
Does any body have any information about Dialogic MSI board workink with
asterisk.
Robin
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Robin van Leyden wrote:
Does any body have any information about Dialogic MSI board workink with
asterisk.
According to this document the MSI model is not supported:
http://www.asteriskpbx.org/index.php?menu=hardware
Keep in mind that the Dialogic drivers for Asterisk are closed source
and cost
Thanks Matthew,
You are the MAN! It fixed the problem.
Richard
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matthew
Sent: Sunday, October 31, 2004 3:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
[EMAIL PROTECTED] wrote:
Robin van Leyden wrote:
Does any body have any information about Dialogic MSI board workink
with asterisk.
According to this document the MSI model is not supported:
http://www.asteriskpbx.org/index.php? menu=hardware
Keep in
mind that the Dialogic drivers
Hello;
I intended to say that certain modifications one brought to the protocol R2 so it can support the E100P degium card and that, for certain country.
I work in Algeria, and ISDN protocol doesnt exploited yet, Therefore I will to make tests with the E100P and R2 modified.
Can someone help
bonjour
je pense vous parler français
sinon pour le pb de la carte digium se resume en l'incompatibilité
avec le R2, tous simplement par ce que le protocol R2 est sous 8bits
or que le EURO ISDN dépasse le s 32 bits, pour cela la seul solution
est que vous utilisé une passerel du genre filtre si
Title: pri usage
Hi,
I have a PRI card. Is it a way to get the usage of the channels in real time and keep in log? For example, through mrtg?
Thanks,
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[EMAIL PROTECTED]
Hi,
I need an advise for a ISDN card for my HomeOffice Asterisk Setup.
Currently I started with a couple of x100p for two anolog lines coming from a
ISDN NT. Works but on bridged calls the sound quality is bad and
distortion, if the call is being routed from the pstn back to pstn on the
Bostjan Repnik wrote:
Im looking for a ISDN card that works under asterisk and supports BRI
line. And I just can`t findit. Momently im using card INTERNAL, but
Im having problems, asterisk on startup when loading modem fails (i4l
driver).
Can you please help me, or point to a www address
Steve,
Thank you for testing our document, and for your valuable feedback. We
are aware that there is still much work to be done, I and apologize that
we have not done a good job of making that clear.
I have answered some of your questions below:
[EMAIL PROTECTED] wrote:
Looks like it's still
Looks like it's still incorrect in the first blue paragraph of the section on
FXO (it's fixed in the second blue paragraph). Also, the last paragraph of that
section twice still calls the channel # 2.
Hi Leif:
I had some similar problems with the docs I found, and struggled for a
while.
Thanks, I'm hoping the result of the grief I'm going through will be a
well documented process :-)
Here is exactly where I'm at:
(power up machine and log in)
my exact /etc/zaptel.conf is at:
http://home.geekster.com/asterisk/zaptel.conf , but the most important
part seems to be
dean collins wrote:
--SNIPOMATIC--
Is there some formal way of setting up a bounty on asterisk wiki? I
pledge $US250 to begin with however I may increase that should someone
show me something fruitful.
--SNIPOMATIC--
Hi Dean,
Just go to http://www.voip-info.org/wiki-Asterisk+bounty, add a page and
Hi everybody,
I've got no success to get a friend in Bogota (Colombia) connected to my
Asterisk. He has got a ISDN Internet connection and the UDP packets will
be fragmented. It seems that the MTU of this connection is round about
400 to 500 Bytes. Therefore most UDP-SIP packages are
Hello Gang,
I'm trying to get asterisk to play with a Lucent iMerge. It seems to
that GnuGK talks to it a bit better. So I'm trying to get this:
PSTN-iMerge-GnuGK-Asterisk.
I'd like to get GnuGK and Asterisk running on the same box. Do they get
in each others way?
Any tricks to getting them
Dear All
My business provides Asterisk consultancy in the UK. I am traveling to
San Diego / Tijuana from the 4th to the 13th and wondered if there were
any fellow Asterisk users who would like to meet for a coffee / drink?
Please email me direct ([EMAIL PROTECTED]).
Regards
John
Robert Berg wrote:
We have had some problems registering the firefly with the Asterisk 1.0.2 it
seams that IAX version doesn't match? How to solve this?
Can you provide a little more information on the problems you're having
with registration? Error messages, from either or both the Asterisk and
I tried to do similar thing with avaya definity.
I end up doing make asteirsk h323 client to avaya deifnity h323
gateway. It worked for my purpose. if you control over iMerge, this
can save a little bit of headache instead of goingthourh gnugk.
good luck.
On Sun, 31 Oct 2004 17:42:32 -0500,
That's what I treid first, but the Lucent iMerge and asterisk don't seem
to play well together.
I have it so calls from PSTN-iMerge-asterisk ring, but I can't get the
call to complete. the iMerge seems to drop the call as soon client answers.
I know someone who got it working with gnuGK
Another idea, not sure if it was stated yet is to just run the ftp
server on a private ip address and/or if you are going to have it on a
public ip restrict by ip address. I run my ftp server on a private ip
which is open to everyone on the private lan and on the public side,
for example I only
Hi all,
Is anyone aware of any simple applications to
display your message waiting status on screen? All I would like is a little icon
in my Windows system tray to tell me I have voice mail - nothing else! I have
tried a few of the "status viewers" in the WIKI page on GUIs, but either I
Dear All
Is there anyone out there who is using a VoiceXML system with Asterisk?
Thanks
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[EMAIL PROTECTED] wrote:
Thanks, I'm hoping the result of the grief I'm going through
will be a
well documented process :-)
We're getting there. :-)
Here is exactly where I'm at:
[snip]
-
fxoks=1 # FXS(green) module in slot 1
fxsks=4 # FXO
Hi all,
The journey is complete, at least for this project.
http://lists.digium.com/pipermail/asterisk-users/2004-October/067289.html
I spent the better part of Halloween putting this together,
I hope its useful, enjoy.
My ftp server is on the fritz so feel free to post on any
Thanks, yes I did read all of your reply (and thank you), and gave those
config files a try. Unfortunatly it didn't work, but I'm getting close
to wondering if it's a hardware issue.
Here is my current situation:
Pictures of the card I received are at:
Title: Message
All
--
System FreeBSD
5.2, Dell PowerEdge 2450
Asterisk installed
from ports (1.0.1)
Only using IAX2
(VoicePulse) and SIP (clients)
Oct 31 18:34:29 NOTICE[165595136]:
chan_iax2.c:2442 iax2_read: I should never be called!Oct 31 18:34:30
NOTICE[165595136]: chan_iax2.c:2442
Hello,
Trying to rewrite my dialplan, and it is a little complex. But my
extensions.conf redirection works, but the referred to contexts result
in invalid extension Please help... I have the extension set to 's'
currently, but originally it was 6044. The change didn't make any
difference.
Could you email me the PDF I am having PASV FTp problems. I have the
same setup. Out of interest which case are you using. I looked at the CF
adaptor you used, but not sure if the Morex 3677 case I am using is high
enough.
Kilburn
JR Richardson wrote:
Hi all,
The journey is complete, at
There is an embedded space in the PDF filename that appears to be
causing ftp to choke. . .
FYI.
Thx.
B.
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files mirrored on voip-info.org here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+embedded+systems
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: JR Richardson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 31, 2004 4:30 PM
Subject:
On October 31, 2004 05:36 pm, Bastian Schern wrote:
I've got no success to get a friend in Bogota (Colombia) connected to my
Asterisk. He has got a ISDN Internet connection and the UDP packets will
be fragmented. It seems that the MTU of this connection is round about
400 to 500 Bytes.
I am new to this VoIP thing and I am looking for a good service
provider for VoIP. I realize that this is a hardware/software list,
but figured that if you are all talking about the equipment, then you
have to know some business class service providers.
Shane Flynn
IT Administrator
Visible
Andrew Kohlsmith wrote:
On October 31, 2004 05:36 pm, Bastian Schern wrote:
I've got no success to get a friend in Bogota (Colombia) connected to my
Asterisk. He has got a ISDN Internet connection and the UDP packets will
be fragmented. It seems that the MTU of this connection is round about
400
Hi Shane,
This type of a request is really meant for asterisk-biz.
However, if you contact me off-list I will forward you our A-Z Wholesale
Termination Rate-Card.
Cheers,
Sahil
Quoting Shane Flynn [EMAIL PROTECTED]:
I am new to this VoIP thing and I am looking for a good service
provider for
Bastian Schern wrote:
Hi everybody,
I've got no success to get a friend in Bogota (Colombia) connected to my
Asterisk. He has got a ISDN Internet connection and the UDP packets will
be fragmented. It seems that the MTU of this connection is round about
400 to 500 Bytes. Therefore most UDP-SIP
On Sun, 2004-10-31 at 15:26 -0500, Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
Robin van Leyden wrote:
Does any body have any information about Dialogic MSI board workink
with asterisk.
According to this document the MSI model is not supported:
Hi, is it possible to change the amount of time it takes asterisk to
pickup an incoming call on a zaptel interface?
cheers
Adam
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Good day all
I want to record all call on my zapvpbinternal channels.
I had a look on the net and and found astGUIclient,I want something easy
and simple that will save it in date/user files.
Please advice
Thanks Altus
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Just an FYI:
If you are *EVER* unsure that mpg123 is correctly installed (correct
verison etc), you can enter the asterisk source tree, and type 'make
mpg123' (without quotes), and mpg123 v0.59r will be download ed,
unpacked, and built for you, and then a simple make install will install
I'm a newbie here. I have a general question that can help drive how
exactly I'm going to get started.
Say I have a single inbound number (1-800-my-number).
When a call is connected on that number, and another call comes in,
will asterisk answer it, will a call waiting signal be triggered, or
On Mon, 1 Nov 2004 15:36:41 +1100, Sophus [EMAIL PROTECTED] wrote:
Hi, is it possible to change the amount of time it takes asterisk to
pickup an incoming call on a zaptel interface?
I presume you are talking about an analog FXO port here. The reason
why it takes Asterisk a while before it
Lister Account wrote:
I'm a newbie here. I have a general question that can help drive how
exactly I'm going to get started.
Say I have a single inbound number (1-800-my-number).
When a call is connected on that number, and another call comes in,
will asterisk answer it, will a call waiting
Sophus wrote:
Hi, is it possible to change the amount of time it takes asterisk to
pickup an incoming call on a zaptel interface?
The command
exten = s,1,Wait,5
would tell asterisk to wait 5 seconds before picking up the line. More
info here: http://www.voip-info.org/wiki-Asterisk+cmd+Wait
Asterisk is working only in Linux? Can not work in
Windows 2000?
Please advise.
Regards
Bilal
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I saw something on the Digium site a few days ago that Asterisk was available
for MS based platforms. Its called AstWind.
http://www.digium.com/index.php?menu=astwind
Cheers,
Sahil
Quoting Bilal Ghayad [EMAIL PROTECTED]:
Asterisk is working only in Linux? Can not work in Windows 2000?
On Fri, 1 Nov 2002 09:46:46 +0300, Bilal Ghayad [EMAIL PROTECTED] wrote:
Asterisk is working only in Linux? Can not work in Windows 2000?
You can have Asterisk on any operating system you like, as long as it
is a proper operating system that actually deserves the name, that is
to say a system
[EMAIL PROTECTED] wrote:
[main]
; 6044 main office line.
exten = 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1)
exten = 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1)
exten = 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1)
exten = 6044,4,Goto(afterhours,1)
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