Hi all, Is there any way to play floating number using asterisk dialplan?
Thanks,Faheem
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
On Mon, Jun 07, 2010 at 01:33:35AM -0700, Faheem wrote:
Hi all, Is there any way to play floating number using asterisk dialplan?
A floating point number has an exponent and mantissa. But I don't
suppose you'de want to know that you have 234 times 2 at the third
power Dollars in your account.
I made some changes to app_voicemail.c and recompiled asterisk. Now my
caller is only presented with the menu-choice I want.
However, the caller can still give another dtmf-input and be taken to
that specific menu.
How can I disable dtmf-input 2,3,4 if I only want the menu behind option
1
Hi!
Hi all, Is there any way to play floating number using asterisk
dialplan?
A floating point number has an exponent and mantissa. But I don't
suppose you'de want to know that you have 234 times 2 at the third
power Dollars in your account.
He probably meant either overlap dial or
This won't help with the exponential number, but is happy for currency or
any floating point number.
Exten = s,1,Set(XFERMAX=100.25)
exten = s,n,SayNumber(${XFERMAX})
exten = s,n,playback(digits/dollars)
exten = s,n,Set(XFERMAXC=${CUT(XFERMAX|\.|2)})
exten = s,n,SayNumber(${XFERMAXC})
exten =
Thanks Danny! It solved my problem.
Faheem
--- On Mon, 6/7/10, Danny Nicholas da...@debsinc.com wrote:
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] How to play Floating point numbers?
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
sean darcy wrote:
Richard Kenner wrote:
Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi.
If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this
is my problem, instead of filing.
I reported another instance of this today and it was fixed in the SVN a
On Sat, 2010-06-05 at 22:16 +0200, Julien Claassen wrote:
But when I make a call;
channel originate sip/iptel-out/e...@iptel.org Application playback
vm/net_ring
The call is onlyleft in state ACK for a while. Then asterisk tells me,
that
it is destroying the sip dialog (long ID)
On Mon, Jun 7, 2010 at 2:15 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
I made some changes to app_voicemail.c and recompiled asterisk. Now my
caller is only presented with the menu-choice I want.
However, the caller can still give another dtmf-input and be taken to that
specific menu.
Hello,
thanks for your help and you time,
I tried your statement in my extension like this:
exten = 777,n, set(TIMEOUT(absolute)=10)
exten = 777,n, dial(iax2/400)
exten = 777,n, hangup()
If peer (400) answers the call the call persists,
Hello everyone!
So now I'm testing with chan_sip and I discovered, that I can make calls,
even if they're only listed as active channels.
But JACK just emmits white noise, with a highger frequency than 8kHz, in my
believe.
A call with app_record shows, that the signal is clear and very
Hello Jared!
OK, now calls go in and out. Even with the syntax:
channel originate sip/mu...@iptel.org application ...
it works. I've tested that with application record.
But, the channel only displays ACK and core show channels doesn't list it as
a call or a processed call afterwards.
On Mon, 7 Jun 2010, Adil Zaaraoui wrote:
I tried your statement in my extension like this:
(context or dialplan would be more correct than extension.)
exten = 777,n, set(TIMEOUT(absolute)=10)
exten = 777,n, dial(iax2/400)
exten = 777,n,
Hello Steve,
Thanks again for the effort.
I tried your dialplan like this in my extention:
exten= 777,1, Goto(absolute-timeout-test,777,1)
[absolute-timeout-test]
exten = T,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten = T,n, hangup()
On Mon, 7 Jun 2010, Adil Zaaraoui wrote:
Hello Steve,
Thanks again for the effort.
I tried your dialplan like this in my extention:
exten= 777,1, Goto(absolute-timeout-test,777,1)
[absolute-timeout-test]
exten = T,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten
I'm succesfully using IAXmodem for faxing (using hylafax) with Asterisk. I
would like a little more control for outbound calls using IAXmodem, but I'm not
sure how to do it. It looks like dialing out over IAXmodem bypasses the
dialplan altogether...can anyone confirm this?
MD
--
On 06/07/2010 01:27 PM, Michelle Dupuis wrote:
I'm succesfully using IAXmodem for faxing (using hylafax) with Asterisk. I
would like a little more control for outbound calls using IAXmodem, but I'm
not sure how to do it. It looks like dialing out over IAXmodem bypasses the
dialplan
Hi Steve,
We are logged using 200 account, it exists in IAX.conf, so user 200 calls 777,
is that right i think?
We did not write dial(IAX2/777|60) but i wrote what you wrote exten= 777,1,
Goto(absolute-timeout-test,777,1)
and we got the the result as shown in the previous post.
User 200, and
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk -g.
But the chkconfig --add asterisk doesn't work :(
I need FOP and this error should go away as it's annoying. I don't see this
on Trixbox, piaf, or Elastix. It shouldn't be on my install either.
Thanks for the input.
On
On Mon, 7 Jun 2010, Adil Zaaraoui wrote:
We are logged using 200 account, it exists in IAX.conf, so user 200
calls 777, is that right i think?
Not call (implying dial()), goto a new context (so we have a clean
environment), where we have a dial() command. You want to put the goto in
the
Back Steve,
Yes you were right, the exten= 777,1, Goto(absolute-timeout-test,777,1)
was not in the context 200,
Now here is the output:
-- Accepting AUTHENTICATED call from 192.168.1.34:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does chkconfig --list asterisk show?
The add command looks in the asterisk script for a line that looks like:
#
On Mon, 7 Jun 2010, Adil Zaaraoui wrote:
Yes you were right, the exten= 777,1,
Goto(absolute-timeout-test,777,1) was not in the context 200, Now here
is the output:
-- Accepting AUTHENTICATED call from 192.168.1.34:
requested format = ulaw,
requested prefs = (),
Hi Steve,
You do really grasp Asterisk, it works, i added transfer=no in iax.conf for
eithet users 200 and 400, and now it hangs up.
For the S option it works too, i write it like this :
exten=888,1, Dial(IAX2/400, ,S(10))
For the script there is a method
Hi,
I'm new to asterisk and have a little trouble in developing my first more
complex dialplan. The basic task is a click to call solution:
- call one number via sip, play some announcements, do cdr etc. and put
the callee into an conference room with music on hold
- call a second number via
Asterisk 1.6
CentOS 5.0
All -
I'd like to offer my users the ability to hangup a call by pressing **. I'm
using an attendant, so when ** is dialled I'd like processing to return to
the attendant so the user can place a subsequent call. I have setup
features.conf to include:
[featuremap]
Steve Edwards wrote:
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does chkconfig --list asterisk show?
The add command looks in the asterisk script for a line
Thanks for the input Seann and Steve. That is insightful. I did run
chkconfig --list asterisk and following is the output:
*[r...@tel ~]# chkconfig --list asterisk*
*asterisk0:off 1:off 2:on3:on4:on5:on6:off*
In file /usr/sbin/safe_asterisk I have priority for asterisk
On 6/7/2010 5:20 PM, bruce bruce wrote:
Thanks for the input Seann and Steve. That is insightful. I did run
chkconfig --list asterisk and following is the output:
*[r...@tel ~]# chkconfig --list asterisk*
*asterisk0:off 1:off 2:on3:on4:on5:on6:off*
In file
chown: cannot access `/dev/tty9': No such file or directory
I had this error on a VPS (virtual server) that did not have access to
tty's. You can take the TTY statement out of safe_asterisk script and
then try it again. I don't have the exact code right now because I'm
on my phone,
I did see the TTY=9 on the third or fourth line but commenting that doesn't
help much. I would really appreciate it if you can send the changes you
made.
Indeed it is a VPS.
Thanks,
Bruce
On Mon, Jun 7, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote:
*chown: cannot access
I'm having a lot of problem with it recognizing oh for zero.
I've tried both o and oh. In one case, I just tried:
$digit = o { out = 0; } | fundamental {out = 2; };
So I gave it a choice that was VERY far away from what I said.
But when I said o o o o o, more than 75% of the time, it
32 matches
Mail list logo
