Displaying 20 results from an estimated 3000 matches similar to: "Directory() Command"
2005 Jan 30
4
Zap channels in AU hanging up on STD pips
Is anyone having/had a problem with a TDM400P card hanging up on STD
outbound calls as soon as the called party answers.
I'm guessing that * is responding to the STD pips in some way.
--
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
------------------------------------------
"When you just want a system that works, you choose Linux;
when you
2005 Jan 27
3
Festival as background
Is it possible to run the Festival command in the same manner as the
Background command so that it can be interrupted by caller key presses?
--
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
------------------------------------------
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose
2005 Jan 14
5
Softphone for Linux recommendation
Can anyone _recommend_ a downloadable OSS softphone that _works_ under
Linux and is compatible with Asterisk.
So far I have tried kphone and linphone and had problems with both, and
I am still waiting to hear back from the X-Lite beta folks.
--
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
------------------------------------------
"When you just
2006 Dec 10
4
X100P clone dial problems.
I'm not sure if I have a configuration problem or not. I am unable to
dial out. When I try to dial in I can hear the phone ring on the
dialling phone but Asterisk does not register anything.
In zaptel.conf I have
loadzone = au
defaultzone=au
fxsks=1
In zapata.conf
language=au
context=from-pstn
When I do: zap show channels I get:
Chan Extension Context Language
2006 Dec 14
4
Zaptel under FC6
Hi, all
I am building a new server. Have installed FC 6 and put in TDM400 card.
Checked out latest asteriusk code, run make install in zaptel directory.
So far all is fine.
Now I am trying to install the drivers.
# modprobe zaptel
FATAL: Module zaptel not found.
Fair enough, no zaptel driver is found on the system.
Is there are any known problems with FC6? I did not have much trouble
running
2005 Jan 09
4
Asterisk Demo
Hi,
I need to setup a demo for asterisk and need some help here please. The demo
is connecting to Asterisk a Cisco 7970 SIP (ver. &.0) and a SIP client on HP
iPAQ via a wireless hotspot. I need to configure both with the same
extension with a shared line like in Cisco CallManager. This way if the
extension is called both iPAQ and the IP phone ring and the user gets to
pick up using either.
2005 Jan 27
2
Q: Can I over-ride the value of ${CALLERIDNAME} ?
Folks,
I'd like to change the value of ${CALLERIDNAME} for incoming PSTN
calls from certain numbers, but haven't found a way that works. The goal is
to provide more informative names on my phones' caller ID displays--e.g., I
would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call
home from my cell phone.
This is what I tried in the context
2004 Dec 18
1
X100P card in Australia
I'm trying to get the X100P card working in AU.
So far I have managed to get it to handle incoming calls from the PSTN
and have managed to eliminate pretty much most of the echo.
My big problem is getting the outbound calls to work. When I get ZAP to
dial out it won't connect and I get what I think is the Congestion
signal - like a busy signal but with what appears to be a 10db
2005 Jan 02
12
phones with two ethernet ports
Hi there, what phones are available that have two ethernet ports?
I want to do some cabling at a new installation and i heard there are
such phones (SIP i guess) out there. That way i dont have to run two
cat5 to the user desktop.
I think 3COM had one but can't find the web site reference for the two
port phone
thanks,
erick
2007 Jan 05
2
SIP/TCP?
I'm still learning some of the basics. Can someone explain in layman's
terms what's the difficulty for Asterisk to support SIP/TCP (and even
RTP/TCP)?
2005 Mar 04
7
Stutter Tone
I think I have something misconfigured regarding voicemails. They work
great, I have this setup:
Sip.conf
[ext1]
Context=phones
Mailbox=201
Voicemail.conf
[home]
201,password,name,email@mail
Voicemail delivery and all works great but when I check sip extension ext1
(analog phone using a Granstream ATA 286), the stutter tone signaling
message waiting does not work.
Anything wrong with
2005 Jan 04
3
Where to start. {Scanned}
Hello All, Yep I'm a newbe.
I'm just started to play with asterisk.
What I have
Redhat Fedora Core 2 (New install)
3 X100P cards.
I installed
zaptel-1.0.3
libpri-1.0.3
asterisk-1.0.3
Where should I start??
--
Thanks, David
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2005 Jan 14
1
iaxComm 0.99pre11 binaries posted to Sourceforge
iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol.
It is distributed as part of Steve Kann's iaxclient library.
I've just posted new Windows, Linux and Mac OSX binaries to sourceforge.
The Windows binary was compiled on WinXP.
The Linux binary was compiled on RedHat 9.
The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4
(Tiger) beta.
2005 Jan 24
3
TDM400 in aging Dell Optiplex
I've got an old Dell Optiplex (Pentium-II, 1998 Vintage) which is
successfully running an X100P card. I'm hoping to upgrade to a TDM400.
Has anybody tried running these cards in old Optiplex machines? I'm not
particularly worried about horsepower - more about the motherboard having
a PCI bus that's able to power up the card...
-Ronan
2005 Feb 15
1
Asterisk "no one is available to take your call"
OK - I can successfully make calls from SIp phone through an asterisk
323 channel to a Cisco Call Manager and out a MGCP controlled gateway.
The problem is that if the call is not answered within ~5 seconds, *
gives the message "no one is available to take your call" and
disconnects the call. If I answer b4 the 5 seconds - everything is good.
Anywhere I need to set to get around
2005 Feb 28
2
Two offices connection
I would like connect two offices where one office have 4 PSTN Analog lines
and another office without any PSTN. Both the offices will have two separate
Asterisk server with TDM400P cards (4 ports FXS & FXO).
My questions is that how to configure Asterisk to forward the PSTN calls
directly
to another Asterisk which has the TDM400P card without pressing the
extension
number.
Diagram like
2005 Mar 09
1
Providing a dialtone
Hi,
I see applications for signalling busy, congested, ringing, progress
etc, which I understand can be provided either in or out of band. But
all I want to do is generate a dialtone. This obviously can only be
done in band.
There is code for generating the tones when you have a physical line,
like the alsa channel, or a zap channel. But I'm just thinking of if
they've selected an option
2005 Jan 31
1
A neat "hot seating" mplementation
Has anyone implemented "hot seating" in any neat way? This where
people can log in to any phone in the company and have their
calls/voicemail come to that particular handset.....
2005 Jan 03
5
Does Digium work on Mondays?
I've been trying all day to reach their techie folks to ask a couple
questions about something I worked all weekend on. Keeps rolling to VM
and the receptionist does the same thing.
Just was wondering if anyone else was able to reach them today.
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)
2004 Sep 20
2
Voicemail Directory
Hi All-
I am running into a small problem trying to implement voicemail
Directory(). I'm sure it is a simple thing, but I can't figure out
where the problem lies.
I can get into the directory without a problem and can look up users
by their last names, however I hit a snag when asterisk says "if this
is the person you are looking for press 1 now". When I hit 1, the
attendant