Displaying 20 results from an estimated 4000 matches similar to: "Why are FXO so expensive?"
2003 Sep 04
2
cisco ATA186 I2 vs I1
Hi,
I saw your posting about the cisco ata186 I2 vs I1 and the simple vs complex impedance.
I ordered a cisco ata186 i2 for use in Canada by mistake, didn't know that I needed the I1
version.
Will the I2 version work in Canada with regular anlog phones, or will I need to change it.
Thanks for your answer.
Samy
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2003 Apr 01
7
Line is stuck off hook...
Greetings,
I am running Asterisk with a T100P and a Zhone channel bank for over a month now. For the most part it works fine but from time to time (about once a week) the system will not let go of a line and will play the greeting over and over. Anyone calling gets a busy signal. If I reset Asterisk everything works fine. Has anyone seen this problem before and fixed it? If so what did you do?
2003 Dec 15
2
Using asterisk as voicemail with SER
Hi,
I'm currently using SER, and the voicemail system there is not stable,
and is lacking IVR.
I'm wondering if I could use asterisk as a voicemail system only,
where calls will get redirected by ser to asterisk and users will be
able to leave a message.
Is this setup workable, and have anybody done that ?
Thanks.
Samy.
2003 Jun 01
2
Any plans for a .....
I saw that Digium now has a 4 port FXS card. Any plans to add a 4 port FXO card?
Gene
2003 May 09
2
All station page and operator console....
Greetings,
Can anyone tell me if there is any equipment or way to configure Asterisk to support station page and if there is an operator console that works with Asterisk? Also has anyone implemented a PA extension using asterisk and if so how was it done?
Thanks, Gene
2003 Mar 03
6
Fax support?
Is there any way to receive and send faxes using a T100 card? If so how is it done?
Gene Kochanowsky
Solution Sciences, Inc.
2004 Mar 31
7
Extension ringing but no ringing sound.
Greetings,
This is probably some configuration issue, but for some reason my system
has stopped playing a ringing sound when an extension is dialed. The
phone rings but there is no ring sound in the ear piece.
Gene Kochanowsky
2005 Aug 01
2
CentOS-3 glut
Does anyone know where I can get glut.h and associated libraries for
CentOS-3?
Thanks
John.
--
John Newbigin
Computer Systems Officer
Faculty of Information and Communication Technologies
Swinburne University of Technology
Melbourne, Australia
http://www.ict.swin.edu.au/staff/jnewbigin
2003 Mar 04
3
Fwd: Re: Fax support?
I can't seem to make the fax detection work. Here's
an excerpt from zapata.conf:
signalling=fxs_ks
group=0
context => guestaccess
channel => 47-48
and from extensions.conf:
[guestaccess]
include => incomingmain
[incomingmain]
exten => s,1,Dial,Zap/1&Zap/9&Zap/10&Zap/11|24
exten => s,2,Voicemail,u7000
exten =>
2003 Nov 10
2
ISDN TBCT....
Greetings,
This may be a bit arcane but does anyone know what the contents of a facility message should be for initiating a TBCT on an NI2 ISDN. I've been trying to get it to work on a DMS100 for the last four months to no avail. The message I am currently sending makes it to the switch but is returned with unknown message. Perhaps someone here has done it before and can help me out.
2003 Jun 01
4
Zapata 3.3v PCI version...
Does anyone know if there are any plans for Zapata or anyone else for that matter to come out with a 3.3v PCI version or PCI-X version of those cards?
Gene
2004 Mar 30
5
Caller entered digits ignored during wait....
Greetings,
Below is part of the contents of my extensions.conf file.
exten => s,1,Wait,1 ; Wait a second before
answering.
exten => s,2,Answer
exten => s,3,ResponseTimeout,10 ; Set the amount of
time the user
; has to
make a selection.
exten => s,4,DigitTimeout,5
2004 Feb 03
2
Pictures of new multiport FXO/FXS from digum
Does the voicetronix card work with Asterisk?
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
woody+asterisk@solutionsfirst.com.au
Sent: Monday, February 02, 2004 11:06 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from
digum
> -----Original Message-----
2010 Apr 05
5
Continuous bothering message -- Remote UNIX connection disconnected
Hi Guys,
i have a small issue but bothering me, after restarting asterisk (version
1.4 running on centos) i have the following message that comes repeatedly
when i am connected to the CLI:
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
does any one know how to stop this or if it's a sign of a
2010 Sep 17
4
Not able to join conference
Hi All,
We are running to a weird problem, we're using asterisk 1.2 as a production
server (I'm wiling to move very soon to more recent version) and our problem
is when somebody try to join a conference he's told that he's the only one
in the conference but in fact there is some 3 or 5 or whatever people in
that same conference, after several tries he can/cannot enter the
2004 Mar 28
1
Programming an unlocked ADSI Astra 390 phone?
Greetings,
I have just purchased several Astra 390 phones ready for asterisk. I
have placed a line with
adsi=yes
in the Zapata.conf file just before
channel => 13
I have also added an extension
exten => 6199,1,ADSIProg(asterisk.adsi)
exten => 6199,2,Hangup
in the extensions.conf file.
When I try to program the phone I get the following:
Asterisk CVS-03/28/04-12:02:10,
2010 Jan 28
3
Dial cellphone from one PBX1 to PBX2? is it possible?
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a
cellphone)
my questions now is : am i gonna be able to dial from an IPphone registered
within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
anybody know
2010 Mar 04
1
InterPBX communication using SIP
Hi Guys,
i am using the following config in pbx1:
register => pbx1:endopass at 172.16.200.175 <pbx1%3Aendopass at 172.16.200.175>
[pbx2]
type=friend
host=dynamic
trunk=yes
sercret=password
context=[default]
deny=0.0.0.0/0.0.0.0
permit=172.16.200.175/255.255.255.128
in pbx2:
register => pbx2:endopass at 172.16.200.176 <pbx2%3Aendopass at 172.16.200.176>
[pbx1]
type=friend
2003 Jul 21
4
Using asterisk for a 911 call center....
Has anyone had any experience using asterisk for a 911 call center? Does anyone know of any reason why it would not be suitable? As far as I know all 911 call routing takes place at the CO switch so a regular T1 line should work fine. I understand that there is support for ACD in asterisk and that is should be possible to implement screen pop (CTI). Any comments?
Gene Kochanowsky
Solution
2010 Aug 28
1
problem after repackaging
Hy,
I had a mistake on a function of a package i have created!
I have solved it and then i repackaged and installed the modified package.
I use to launch R from Excel!
And so when i launch R, and next call my function from the workspace, i
still find the problem on my function.
And when i read on my workspace, the source code of my function, i find the
old version of my function (the one from the