We got version
Asterisk CVS-HEAD-09/01/04-11:36:41
and all works well for us with the voicetronix card
Then we upgrade to a newer version of * and it did not seem to carry
over the DTMF signal out of the PSTN
Voicetronix fixed this so we got the new version of asterisk and it does
not allow us to
it is a good idea when users get the cvs version.In this way
when there are more a chance of someone discovering a bug/problem.
On Fri, 2004-12-10 at 17:39, Bob Goddard wrote:
On Friday 10 December 2004 14:47, Altus Snyman wrote:
Please do not top post.
We got version
Asterisk CVS-HEAD-09/01/04
Good day all
We have Grand Stream BT-100 phones
The transfer button work well, for blind transfer
What the users want to do is, when a call comes in and asked to be
transferred to another extension,for example 100,they 1ste want to speak
to exten 100,then have the option transfer or not to
-
From: Altus Snyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, December 09, 2004 4:21 PM
Subject: [Asterisk-Users] BT-100 Transfer!!
Good day all
We have Grand Stream BT-100 phones
The transfer button work well
I want to use it in my pbx as a bri card for incoming and outgoing calls
in asterisk
How did you get it working with asterisk,drivers ens.
Please Help
Andrew Kohlsmith wrote:
On December 8, 2004 02:40 am, Altus Snyman wrote:
Is there someone that's got asterisk working well
Good day all
We have a Mitel 3300 connected to a grandstream handytone 486
These is a conference unit,one big speaker phone,my question is how to
make a conference call using asterisk
Other phone has the conference button on so if you press it you can call
someone else and all can talk together
Sorry its a Mitel 5305
Altus Snyman wrote:
Good day all
We have a Mitel 3300 connected to a grandstream handytone 486
These is a conference unit,one big speaker phone,my question is how to
make a conference call using asterisk
Other phone has the conference button on so if you press it you can
Good day all
We got the cvs yesterday,and it seems as if transfer does not work.We
are using mitel 52205055 and the Grandstream bt-100,using the transfer
buttons.
If you transfer it just goes to the next step?
please Help
Thanks
Altus
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Good day all
Is there someone that's got asterisk working well with a A101/E1 card
Apparently they don't have RBS support?
Please advice
Thanks
Altus
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What about the mitel 5220's buttens?
Tracy R Reed wrote:
On Sat, Dec 04, 2004 at 08:03:03PM -0400, Cian O'Sullivan spake thusly:
She is an older lady and does not want to use a web interface. Any
suggestions?
Give her a Snom or Polycom phone which does have this
Do you have the callerid thing in sip.conf? And the setting on the phone
"User ID is phone number:" = no
Myne works
Rodney Acosta Coya wrote:
hi all,
i have some grandstream bt100 registered with asterisk
when a extension receive a call display its own number
but i need to diplay the
Good day all
We have a voicetronix openline4 card
If someone calls in from the outside the pstn and into the system and
hangsup asterisk does not deteck the hangup
any Idea why
please Help
Altus
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How and what and where?
Sorry I'm a bit new to asterisk and programming
Thanks
Altus
el Flynn wrote:
Altus Snyman wrote:
Good day all
We have a voicetronix openline4 card
If someone calls in from the outside the pstn and into the system and
hangsup asterisk does not deteck the hangup
any Idea why
There is a version 18!
Michael Nolan wrote:
On Wed, 24 Nov 2004 11:02:36 +, Bob Goddard [EMAIL PROTECTED] wrote:
On Wednesday 24 November 2004 10:39, Simon wrote:
I've searched for a few days now without finding an answer. The
release notes for
Good day all
How do I dial from the cli
It says dial [number] but that doesnt do anything?
Thanks
altus
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What phones are you using
I know the grandstreams you have *70somethings in,bin a while
The phones have these settings,Mitel 5055 you can do it on the phone web
pages
ismaelg wrote:
Hello all,
I want to setup Asterisk to forward a call if the dialed extension is
busy. I do not want to wait on
Good day all
I want to tell asterisk that it should hangup a channel in a certain step
For example:
exten = s,5,Dial(SIP/302,25)
exten = s,6,Hangup
exten = s,7,Hangup(SIP/302)
What happens is that if someone calls into the pbx and hangs up before
it gets answered it still keeps on ringing on the
Good day all.
I have a small problem
When someone calls in from the outside,dial the extension of the
internal sip phone,and the hangs up without getting any response,the sip
phone will keep on ringing?
It show that is hangs up on the Zap/vpb channel but the connection
between asterisk and sip
Good day all
I upgrade my asterisk and the vpb driver to the latest
I used all my previous, working, config files over.
Every thing works well but for 1 thing,playing DTMF when making a
outbound call
If I call a external number on my phone and another pbx answers and I
have to press a number it
I installed the new version of asterisk
But the other probelm I got was,were are using the voicetronix cards,so if
you go and put
ignorepat = 0
exten = _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _0.,2,Monitor(wav,${CALLFILENAME},m)
exten = _0.,3,Dial(vpb/1-3/${EXTEN:1})
exten =
Good day all
What do i need,and need to do,from the beginig to get the cdr into a
database
I installed asteris,mysql and mysl-dev..
I got the asterisk-addons
I made a make and it said it could not find asterisk.h
so I copyed asteris.h from my installed file to /user/include
I then followed theis
Good day all
I have my extensions.conf configured so that it waits 8s the answers
with a message saying press 1 for... and 2 for..
How do I tell it then that if the did not press anything to should go to
the operator.
And/Or if they did not press something it will play the message again
And/Or
Good day all
We have the Grandstream phone.We configured sip.conf to use libc and the
phone as well.But now,where in alaw it worked,if you enter the voicemail
it doesnot recognize the number you type in,in other workds,if it asks
for mailbox and I type in 405 and check asterisk it says no
Good day all
I have my extensions sorted out nice but I need some help with more advance
config.In short myne looks like this
[company1]
exten = s,1. plays the message,saying 1 for sales 2 for accounts ens
.
.
.
exten = 1,1,Dial(SIP/40615)
exten = 1,2,Dial(SIP/403,15)
Good day all
We are running x-lite as a sof client and using the zaptel cards
Each time I make a call out I get a big echo but when I get a call in there is
no echo?Why is this
Please Help
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Good day all
I added the music on hold entry in vpb.conf and commented out default line in
musiconhold.conf.
Asterisk starts up with the default mp3 but as soon as I remove it and add my
mp3 it just doenst start up and gives a broken pipe error?
Please Help or advice
Thanks
ALtus
In the howto it tells me I should strip the ID3 tags
How do I do that?
On Monday 13 September 2004 14:24, Andreas Roedl wrote:
Hello!
Am Montag, 13. September 2004 13:53 schrieb Altus Snyman:
Good day all
I added the music on hold entry in vpb.conf and commented out default
line
Good day all
I'm just wondering for interest sake
I have a box,hostname=myname.co.za,running sendmail
If I send mail to [EMAIL PROTECTED] it try to deliver to the box,witch
does not have the mail box.How do I tell sendmail that it should send mail to
myname.co.za's mailserver.
I know how easy it
Good day all
I'm interested in video on asterisk using SIP and windows clients
Now I did my research on http://www.voip-info.org/wiki-Asterisk+video
I have a few question:
*On the page they say you need the H.261 H.263? codecs,are these compiled in
by default or do I need to do something special
wrote:
On Fri, 2004-09-03 at 10:56, Altus Snyman wrote:
Good day all
I'm interested in video on asterisk using SIP and windows clients
Now I did my research on http://www.voip-info.org/wiki-Asterisk+video
I have a few question:
*On the page they say you need the H.261 H.263? codecs
Good day all
Is there anyone who has experience with ISDN BRIDDI?
I want to know if asterisk can distinguish between the different numbers?
I want each number to play a different intro/answering message?
Please Help
Thanks
Altus
___
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I only did this with ports and the using context how do I do the number
I want each number for a different company?
On Thursday 02 September 2004 15:10, Darryl Ross wrote:
On Thu, 2 Sep 2004 14:51:10 +0200, Altus Snyman [EMAIL PROTECTED]
wrote:
Good day all
Is there anyone who has
Good day all
I'm new to the whole pbx thing.I've setup 2 servers with voicetronix card!
Each card's got 4 ports.Ive configured it so each port is for a different
company,so in other words if a call comes in on port 1 it plays company 1's
welcome message ens..I did this with context in vpb.conf
Good day all
I've tried my iax conf and I'm struggling.So I want to know If someone
else got this working and if they can pleas send my their configs
I have to asterisk server,in different tows,both offices connected wit a
direct line so both servers are on the same network running SIP.Each
town
Good day all
I'm trying to configure 2 asterisk servers running sip to connect with
each other with iax so both sip extensions can dial each other
I'm using this webpage but I'm a bit stuck
each time I try to dial the other server's sip extension it says trying
and then just gives a busy tone.In
Good day all
I want to know how to configure asterisk so that for instance if you
press *5 it will pickup any ringing(unanswered) calls.
My problem is this,at lunch time a bunch of people go out for lunch and
when a call comes in it just ring and go threw the whole step.
I want someone,whoever is
and in a vpb card?
Thanks
Altus
On Wed, 2004-08-18 at 13:50, Andrew Kohlsmith wrote:
On Wednesday 18 August 2004 04:54, Altus Snyman wrote:
I want to know how to configure asterisk so that for instance if you
press *5 it will pickup any ringing(unanswered) calls.
Yup this can be done
sorry,using the vpb.conf so card like voicetronix openline 4 card.
Sorry my bad
On Wed, 2004-08-18 at 14:32, Andrew Kohlsmith wrote:
On Wednesday 18 August 2004 07:51, Altus Snyman wrote:
and in a vpb card?
Pardon my ignorance, but what's a vpb card
Good day all
I'm still struggling with getting asterisk to hangup.
If I make a call out threw my vpb pstn and the person on the other line
hangs up 1ste it still shows the line is busy!Only after I hanged up it
will show its still open?
Why?
Please le me know
Thanks
Altus
Good day all
We have a voicetronix openline4 card.Asterisk is configured for sip with
all the extensions and allall.
I can call out and internally,to dial out I have to dial 0...
My problem is with incoming calls
If I call my external PSTN number,asterisk answers with the default
message and if I
Good day all
I'm using sip protocol and a openline4 card.If I dial out of the pstn
and hangup a answered call it does not disconnect the connection.It
shows there is still a call on the external phone I called but on my
side its says i'm not connected.We are using x-ten soft phones
Can someone
I'm getting this in debug mode
vpb/1-1: chanreads: Couldnt get lock on owner channel to send frame!
vpb/1-1: chanreads: Finished cycle...
vpb/1-1: chanreads: Starting cycle ...
vpb/1-1: chanreads: Checking bridg
On Fri, 2004-08-13 at 12:12, Altus Snyman wrote:
Good day all
I'm using sip
Good day all
IS there a way to personalise the voicemail message when you leave a
message?
Thanks
Altus
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Good day all
I have figured out most/basics of asterisk.I went with sip and made my
own sip.conf and extensions.conf
No I have 2 servers running sip in different towns.Both is connected
with static ip so thats fine,but now.
Lets say someone want to call someone else in the other town.How do I
get
:03:59 +0200, Altus Snyman [EMAIL PROTECTED]
wrote:
Good day all
I have figured out most/basics of asterisk.I went with sip and made my
own sip.conf and extensions.conf
No I have 2 servers running sip in different towns.Both is connected
with static ip so thats fine,but now
What about outgoing
How do I tell it all sales,sip 100+, to go out threw vpb card's channel
and all admin,sip 200+ to go threw zaptel?
Thanks for the help so far
On Tue, 2004-07-27 at 16:59, Seth Remington wrote:
On Tue, 2004-07-27 at 09:48, Altus Snyman wrote:
Ya but the one is zaptel nd
Good day all
We have a zaptel card in my pbx system.I configured sip for different
department,100+ for sales.200+ for admin.
Now I have added a openline 4 card.My question is,can I configure each
card for different departments,for example,all calls coming in on zaptel
will say welcome to sales
Ya but the one is zaptel nd one voicetronix so it uses vpb.conf for
example sales
On Tue, 2004-07-27 at 15:39, Seth Remington wrote:
On Tue, 2004-07-27 at 06:07, Altus Snyman wrote:
My question is,can I configure each
card for different departments,for example,all calls coming
Good day all
How do I get my asterisk and sip to use the password.I'm using x-lite.If
I use just the username and no password it still logs on?
Here is my sip.conf entry?
[101]
type=friend
callerid=Test User 101
context = test_1 ; Default context for incoming calls
username=101
Good day all
Is it possible to transfer sip calls?And how?
I saw transfer in iax.conf?
Thanks
Altus
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1 solution 4 yout workstation:
Linux Linux Linux Linux Linux Linux Linux Linux !!!
:-)
On Tue, 2004-05-11 at 05:53, John Fraizer wrote:
tmpm wrote:
Of course, and I suggest a firewall as well, but its NOT going to do
anything for a purloined email some infected machine in
Good day all
I want to put the openline4 card into a box that will support 3
different companies
I read the caller ID id fixed but now HOW DO I:
If a call come in for 12345 it plays company 1's welcome message
If a call come in for 98765 it plays company 2's welcome message
ens..
Does This make
The thing is its 3 companies,3 different number 3 different lines.
I know you can sort it with source number(That old girlfriend thing) but
what about destination number,can you get it
On Fri, 2004-04-23 at 10:19, Jeremy McNamara wrote:
Altus Snyman wrote:
Good day all
I want to put
But who do I differentiate between the different number,how do I say: if
a caller calls 1234(the destination) do:
[company1]
exten = s,1,Answer
exten = s,1,Playback,company1-welcome
ens.
On Fri, 2004-04-23 at 10:44, Jeremy McNamara wrote:
Altus Snyman wrote:
The thing is its 3 companies,3
Good day all
I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
But in the pdf howto they speak about a swb.txt and I dont have that
file
Thanks
Altus
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Thanks
On Thu, 2004-04-22 at 09:16, Pertti Pikkarainen wrote:
I'm so sorry. The file is now there.
Please download it.
Thanks !
Best regards Pertti
Altus Snyman wrote:
Good day all
I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
But in the pdf howto they speak
-04-22 at 09:16, Pertti Pikkarainen wrote:
I'm so sorry. The file is now there.
Please download it.
Thanks !
Best regards Pertti
Altus Snyman wrote:
Good day all
I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard)
But in the pdf howto they speak about a swb.txt and I
.
But to be sure later you can easily fix that.
The error is due to a typo in the end of the file
Run the first GRANT command again with 'asterisksettings' and not
'asterikssettings'
I just fixed the download file.
Best regards Pertti
Altus Snyman wrote:
Is this error ok? When I
, 2004-04-22 at 09:31, Altus Snyman wrote:
Is this error ok? When I insert txt file into the db,Im loged in as
postgres
CREATE TABLE
INSERT 16984 1
CREATE TABLE
CREATE TABLE
INSERT 17003 1
CREATE TABLE
CREATE TABLE
CREATE TABLE
INSERT 17020 1
INSERT 17021 1
NOTICE: CREATE TABLE
Good day all
Is it possible to run asterisk and sip without any
cards,(t100,voicetronix)
Just a plain linux server,running mail and web, and add asterisk
At the moment they are running msn?
Tanks
Altus
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Good day all
Did someone get the new ver0.5 flash panel working
Is it suppose not to show who the caller is calling,like on ver0.2?
And how do I change the language
Thanks
Altus
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is wrong with the postgre access rights.
Did you remember modify
/usr/local/pgsql/data/pg_hba.conf
If a new start doesn't help
please, send me $CATALINA_HOME/logs/catalina.out
Best regards Pertti
Altus Snyman wrote:
It comes up with the index page but when you login with admin,admin
Good day all
I'm still looking for a SIP client that will work on fedora core 1?
Thanks
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Do you have a copy for me,the page seems to be closed and it redirect me
to http://swpat.ffii.org/ and I cant read that
Thanks
On Wed, 2004-04-21 at 09:16, Tracy R Reed wrote:
On Wed, Apr 21, 2004 at 09:20:54AM +0200, Altus Snyman spake thusly:
I'm still looking for a SIP client that will work
Yes me to,how do I contact you
On Tue, 2004-04-13 at 13:27, ePyron Felix Deierlein wrote:
Hello Pertti,
we would be interessted to, if you could send further informations...
Thanks
Regards
Felix Deierlein
[EMAIL PROTECTED]
-Ursprüngliche Nachricht-
Von: [EMAIL
Good day.
I'm looking for a sip client 4 fedora???
Frdora?
Thanks
Altus
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Good day all
I'm looking for a GUI/Web interface for Asterisk.
What I need it for is to see who's line(SIP) is busy work?
Something like a switch board?
Please give me some info?
Thanks
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please let me know if anyone get this..please
On Thu, 2004-04-08 at 13:21, Joe Dennick wrote:
I'm still having problems being able to get the Transfer function to
work. I enter the correct password, but still can transfer or end calls
with the Flash Panel. Any suggestions?
Joe
ok this is what I did
I moved all to my /var/www/html/control. did the changes is my files and
used the copy of manager.conf. I started asterisk and did
/var/www/html/control/op_server.pl and pointed my browser to
192.168.0.1/control/html ... had the same problem. Then I went and set
debug to 1
Good day all.
I need a windows client that can transfer calls from 1 user 2 another
with a nice GUI for non PC iterated people
Thanks
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To
http://sip.house.com.ar/operator/
On Thu, 2004-04-08 at 16:01, Steve Foy wrote:
Hi again :)
Can you give me a URL for the software you mentioned?
Cheers,
Steve
On Thu, Apr 08, 2004 at 09:45:47AM -0400, Jain, Sonal wrote:
I installed the flash operator panel and I also installed the
Good day
When I call in from the outside(PSTN),my box answers with the demo,but
it is very soft,and when I dial the extension to my client the
connection is very soft as well,Please help
Thanks
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Good day all
So I've installed asterisk with my openline4 card and I've setup sip and
I can do sip on the local network,we are using soft clients,x-lite.
But...
When a call comes in from the outside(PSTN) and the you dial the
extension it forwards the call the the client and you see incoming call
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Good day
Does Asterisk work with the Voicetronix Openline4 cards?
Thanks
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The thing is,Im not the sharpest tool in the shed, and I really need
help setting it up.I've installed Asterisk but thats how far I'm
getting,would you please Help me,Please
On Mon, 2004-03-29 at 14:33, michiel betel wrote:
Altus Snyman wrote:
Good day
Does Asterisk work with the Voicetronix
Good day all
Now
I want to install a complete pbx system on my linux box with windows
clients.
Now I have the a openline4 card and 4 lines,but what software do I
need,Asterisk running on the server and?and what for the clients,I
see in the config there is a sip provider config??
Thanks
Altus
Did you get it working,its been 7day and 7 nights and I cant dial out?It
receives the demo call but thats it.Please help me
On Mon, 2004-03-29 at 14:33, michiel betel wrote:
Altus Snyman wrote:
Good day
Does Asterisk work with the Voicetronix Openline4 cards?
Yes, see: http
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