harry gaillac wrote:
Hello,
I try to compile zaptel .
I installed kernel-sources but when i run :
make linux26
/
serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make
linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE
kchase wrote:
Can anyone tell me how to recover an Asterisk password if it is
forgotten or do I have to do a complete re-install.
Which password?
Do you have the SSH password for the [EMAIL PROTECTED] server?
David
Kris
rebuild zaptel driver after kernel update
asterisk -r Asterisk CLI
yum -y update Get latest patches for CentOS
You would run
passwd-maint
Hope that helps.
David
- Original Message -
From: David Uzzell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
trixter aka Bret McDanel wrote:
On Wed, 2005-11-16 at 13:22 +1300, Richard Malcolm-Smith wrote:
As far as I was aware a license was only required in contries that had
software
patents, I know that there arnt here so I am just seeking clarification if
thats
all there is to it.
I say just
trixter aka Bret McDanel wrote:
On Wed, 2005-11-16 at 13:21 +1100, David Uzzell wrote:
trixter aka Bret McDanel wrote:
On Wed, 2005-11-16 at 13:22 +1300, Richard Malcolm-Smith wrote:
As far as I was aware a license was only required in contries that had
software
patents, I know
Mir wrote:
Thanks for your suggestion.
Unfortunately, it didnt change anything, A can still not hear B, but B
can hear A, strange..
I had the same problem with one of my IAX providers in AUS.
Both ends turned of trunking and all was fine with the world again.
Not sure what was the
Ok I upgraded tonight a server from CVS in Late NOV to one just
downloaded tonight.
It all runs up OK and I can contact it from my ATA 186 using g729a codec
and that all works fine.
What I am having trouble with is connecting through IAX ATP.org.au in
AUS to my server.
The connection
with g729a
that it would be either IAX2 problems or iblc codec problems :(
Has anyone got any advice?
Thanks
David
David Uzzell wrote:
Ok I upgraded tonight a server from CVS in Late NOV to one just
downloaded tonight.
It all runs up OK and I can contact it from my ATA 186 using g729a codec
Kamran Ahmad wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk
Bharat M. Sarvan wrote:
Hello Everybody,
This is Bharat here. I am on the way of
learning Asterisks, and I just wished to know how I go about if got to
write dailplans for outbound calls and inbound calls. If you could
provide me with a simple example, I could get
Geoff Nordli wrote:
Hi Everyone.
On the Linux 2.6 kernel do I need to recompile the kernel in order to
compile the zaptel modules?
Using Mandrake 10.1 which runs standard a 2.6 kernel I did not have to
recompile the kernel to get them working.
cheers,
David
Thanks,
Geoff
Brett, Gary wrote:
Hi there
Just a quick question. I have been playing around with asterisk CVS-1.0.02
on fedora core 1 (2.4 kernel) and I would like to have a look at asterisk v
1.0.6 but am still a little uncertain which linux kernel is best to run on
?, can I use Fedora Core 3 (is it the
I have this exten code to dial out to a specific group of numbers.
exten = _10X,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:1})
I want to be able to have one line to handle generic calling so we could
dial say 10 or 10X or 10XXX or something in between.
Does any
Does anyone know how long the orders take?
I ordered some a couple of days ago and it said normally 24hours, and I
am guessing that the weekend cause's some delays but it did not say
anything abouy that.
Any one got any ideas on how long generally over the weekend it takes?
Thanks
David
I have seen the list of codecs for the ATA 186's but not sure if it was
100% or not.
I want to know really is it possible to run GSM or ilbc on them or is a
G729 lic the only way to get a low bandwidth codec?
This is the list of codecs that I have seen.
RxCodec and TxCodecConfigure the codec
Martijn van Oosterhout wrote:
On Mon, Feb 28, 2005 at 10:04:27AM +0100, Dave Cotton wrote:
On Sun, 2005-02-27 at 17:54 -0500, Robert Webb wrote:
This is getting VERY annoying.
Is there anyone in here that has access to the list administration to
delete the user below???
Pray tell me why. The
Duane wrote:
For those interested, I'm giving a talk about VoIP/enum.164/asterisk
tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS
build #2, 4th floor, room 10.
Sorry for the late notice, it didn't occur to me that there might be
people on this list interested and able to
could possibly
take advantage of some of the bandwidth saving features in IAX for your
connection.
Just a thought and I don't know how that will work with your operation
but that is what I thought might be an Option.
David Uzzell
___
Asterisk-Users
Jonathan Lin wrote:
Hi All,
I am having trouble getting speex to work on asterisk. I downloaded
1.0.4 from speex.org, download libogg from vorbis. ./configure, make and
make install for both and then recompile my asterisk 1.0.5, yet I am getting
error when trying to load codec_speex.so.
Ok I have a question. Seen it come and go around the mailling list for a
while but never really seen an answer that seems to sort it out.
What is needed is some interface from * Mobile Phone Mobile Network
Service.
At this point all the providers in AUS that I have found are charging a
[EMAIL PROTECTED] wrote:
Hello All,
Can someone please tell me about Zaptel?
Is it only needed if you are going to have an interface card like TDM400P
installed on the Asterisk server?
Do you really need it if you do not have the interface card?
You don't need to load Zaptel if your not going to
and it does poll the * server but does not work to use
and errors in sip.
Followed the instructions on the wiki page for them and it still wants
to be a pain :(
Other problem is that it is in Denmark and I am in AUS :) so timming is
an issue.
Any advice would be appreciated.
David Uzzell
Ralph Green, Jr. wrote:
On Mon, 2005-02-14 at 13:08 +0100, Jens Kübler wrote:
Maybe you wanna check out the softphone zip4x5 made by Zultys.
It's the software which is used by the same hardphone.
Howdy,
Do you use this product and do you have any relationship with Zultys?
It looks interesting,
Nitesh Divecha wrote:
Hey All,
Just finished installing Asterisk and configured all the necessary
parameters to start.
I cant seem to find the Meetme application in my asterisk directory.
I downloaded asterisk from CVS and installed it and all my Snom phones
are working and voicemail too.
Jens Vagelpohl wrote:
On Feb 11, 2005, at 16:28, [EMAIL PROTECTED] wrote:
Extreme N00b, I am getting the error message a target does not exist
when
running the make install inside the zap directory, probably pretty
common,
possibly a package I didn't install, just need some insight on it. The
Ok I have a challange that I can't seem to find a way to fix it.
My Voicemail in * timesout after 30secs without fail everytime no matter
what I do.
I have incomming calls comming in through Freshtel IAX2, if it goes to
SIP extension when it is online it can hang on for what ever time the
call
in there before the VM or would I?
Thanks
David
On Tue, 08 Feb 2005 11:34:30 +1100, David Uzzell
[EMAIL PROTECTED] wrote:
Ok I have a challange that I can't seem to find a way to fix it.
My Voicemail in * timesout after 30secs without fail everytime no matter
what I do.
I have incomming calls comming
Walter Klomp wrote:
Hi,
I'm sorry to bring this up again, but I have been googling forever and
whatever solutions are offered don't work for me.
I am using x-lite (the latest build) and trying to use Speex.
When I do call from the x-lite to another SIP phone or PSTN (through Cisco
gateway) My
Now I have searched around and not seen anything to do this.
I want to in remote locations were we need to have single or 2 PSTN
lines for in dial as little hardware as possible and as stable as
possible so that they will operate without user intervention.
What I want to do is be able to take a
mohammad wrote:
Hi;
I downloded asterisk CVS-HEAD 12/20/04 but I canot see parking.conf in
/etc/asterisk.
There is no longer any parking.conf. It is now know as features.conf and
you can find all the info about it at
I have been tring to send email to the List for the last day or so and
have not seen it come back on the list and have not seen any reponse's
to my email so I am unsure if it is making it to the list.
And I have also been seeing some of the same emails over and over again.
The list does not
I am playing with the dialplan to get it working and I have a challange
with this error. I can't find what it means on the wiki :(
Any sugestions would be helpful at being able to forward it to the SIP
phone if it is online and avaliable but then let that fail and drop into
voicemail if it is not
different SIP
phones both soft and hard to do testing.
why not slim down your peer entry a bit?
ie:
[800]
type=friend
username=800
secret=password
callerid=800
host=dynamic
dtmfmode=inband
mailbox=800
nat=yes
canreinvite=no
David Uzzell wrote:
I am playing with the dialplan to get it working and I have
Sorry for replying to my own mesg but I have more info.
David Uzzell wrote:
Matt Hess wrote:
is this current cvs or something? It looks completely abnormal for
stable..
Ah sorry it is CVS 12/12/04
I have just this min downloaded the latest CVS as of about 20mins ago
and compiled etc
I am playing with the dialplan to get it working and I have a challange
with this error. I can't find what it means on the wiki :(
Any sugestions would be helpful at being able to forward it to the SIP
phone if it is online and avaliable but then let that fail and drop into
voicemail if it is
http://www.itnews.com.au/newsstory.aspx?CIaNID=17357eid=1edate=20041221
Did not know how far and wide this was but I thought it might be of
intrest to people on here.
QUOTE
The Asterisk PBX runs on Linux and provides three VoIP protocols. The
software PBX provides voicemail services with
Martin List-Petersen wrote:
On Sun, 2004-12-19 at 02:11, David Uzzell wrote:
Then the other thing if mem serves me you are running 2.6 kernel so why
not run ztdummy? With the 2.6 kernel this does not require any
specialist Hardware or anything!
Sorry, but maybe you should have read his posts
Brian West wrote:
I feel this is a slap in the face for those of us that have been here and I
don't feel I should HAVE to pay to be certified... I think me and MANY
others are about to walk out of the project over this. I have already
spoken with many people that are close to the project.
Nathan Alberti wrote:
I am having problems getting incoming caller id to work on a Telstra
Onramp 10.
I have changed /DEFAULT_CIDRINGS 2/
Is there something i'm missing ?
My Cisco 7960 just shows asterisk
Thanks,
Nathan
SNIP
linux*CLI show channel Zap/2-1
-- General --
Name: Zap/2-1
Keith O'Brien wrote:
Thanks. On a related note. The problem that I am troubleshooting has to do
with an IAX connection to TELIAX. Outgoing calls are perfect. When I have
incoming calls they are very crackly and break up. I have checked the
jitter buffer and it is not overrunning so it
Ok I have spent the last week working on getting my small PBX to work.
I will in the end only have 4 SIP extensions being either softphones of
IP phones. Currently only 1 SIP config for testing.
And at the this point it should be all fairly easy with all inbound and
outbound to PSTN will be
Jean-Michel Hiver wrote:
I've been setting * at home just to train myself with it. Here is what I
have:
- IVR menu
- music on hold / transfer
- voicemail
- transparent Zap or IAX routing
- I can call home, dial a pin and make long distance call through IAX
It would be great if you could share
Is it just me or are there problems?
The list has just shutdown over the last 24 hours :(
David
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Does this sudden rush of email mean we are all back online?
David
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Just working on Configing up Voicemail and now that I have got it
working and configed and answering the way it should be I have another
challange.
on the * CLI I get this
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/6001/INBOX/msg format:
Ok I have an * server finally setup and acepting calls from freshtel and
I am VERY impressed at how well the Freshtel.net service works but thats
another subject :)
I have it all setup so that I can Dial my DID number on freshtel and
that gets set to my * via IAX.
At the moment I have the
Well here is something simple, well I think it is for the smarty's out
there :)
I got a connection to freshtel and want to get the iax working.
I have config'ed up iax.conf with the register line and get in return in
the cli
-- Registered to '202.168.7.130', who sees us as 203.29.98.221:4569
Has the wiki died or is it just my routing to the wiki from Australia?
I have not been able to connect to it for the last hour or more :(
David
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To
190.263
ms 185.715 ms
17 * * *
18 * * *
19 * * *
Me hopes it is not down to long am in the middle on tring to config my
now working * server :(
Oh well at least it is not just me.
Thanks
David
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af David
James H. Thompson wrote:
Commpartners (who provides hosting for voip-info.org) is doing a network
upgrade tonight.
AH That would explain it :) Just happens to be when I want to work how
to config * :(
Oh well will just have to wait.
To bad we could not mirror the wiki around the world. I am
dean collins wrote:
Michael,
Xorcom can allow ssh. You didn't read the instructions properly (lord
knows I didn't the first few dozen times).
When you insert the disk for the first time instead of typing linux or
pressing enter to start the install type expert
This will halt the installation at
Michael Graves wrote:
On Mon, 29 Nov 2004 10:09:26 +0200, Gilad Ben-Yossef wrote:
Alex Brecher wrote:
Which Distro is the most commonly used distro with Asterisk please ?
I don't know which is most commonly used, but I can tell you which is
the easiest to install if you're going to install the
Please 1 is enough.
Upto 4 so far :(
David
Voip Business wrote:
Hello Lists
I need help to setup a GNUGK with an Asterisk my needs are simple
I have:
billing system compatible with GNUGK
SIP ATA devices for origination
h323 and SIP termination services
1 linux box (can be 2 in same datacenter)
I
I am after something similar.
I want to be able to use 2 bonded ISDN BRI's and I am not sure what
hardware will run with asterisk?
Anyone got any ideas?
Cheers
David
Miroslav Nachev wrote:
Dear Bartosz,
Try this: http://www.junghanns.net/asterisk/page17.html
quadBRI PCI ISDN EUR 600,-
Ok I have searched the Archives and can't seem to find anything that
would give me a hint on if it is possible or not.
What I want to look at having a Small Home Office setup were I can use
the 1 BRI for both DID and Internet at the same time.
Is it possible to use something like a FRITZ! ISDN
Thanks for your advice also Colin, I would like to stay away from POTS
cards in the server but it is an option if all else fails.
Philipp von Klitzing wrote:
Hi!
What I want to look at having a Small Home Office setup were I can use
the 1 BRI for both DID and Internet at the same time.
Is it
Low, Adam wrote:
I've done a fare amount of analysis on codec bandwidth requirements and you should remember that you typically will require more bandwidth over ADSL than you would over any other technology. I estimate a requirement of around 108Kb (on the wire) per G.711 channel rather than 86kb
Dan wrote:
Hi,
...
I won't bother with any of that - purchase a Nokia Premicell (or other
manufacturers similar item). This device takes a normal GSM SIM card and
then presents a normal PSTN line interface - plug that into your normal
Asterisk PSTN line card - job done.
A PCI and/or USB
to find an USA reseller.
OK think thats all! Thanks Guys doing some great work with what looks to
be some very GREAT software!
Thanks in advance for your help!
David Uzzell
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