On Thu, 15 Jan 2004, Steve waxed:
On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote:
sounds like one of those pesky auto dialers the simpsons make fun of.
It sure does...
The AT-5000 was Prof. Frink's first patent, and it was
designed to alert children of snow days and such. I
On Thu, 15 Jan 2004, mattf waxed:
8's
There is a group of Asterisk users that decided to modify the code of
Asterisk to try to make it a predictive dialer, called shady_dial I believe,
but I haven't heard anything about it lately.
http://shadydial.sourceforge.net/
Lots of recent updates
On Sun, 18 Jan 2004, Ulexus waxed:
On Sunday, 18 January, 2004 02:04, Ken Alker wrote:
Assuming the price of an ADSI screen phone (say, Aastra 390) was the same
as an IP screen phone (say, Cisco 7960) and someone was setting up an *
server for their 20 employees (each of whom would have
On Mon, 19 Jan 2004, Ted Cabeen waxed:
Andrew Kohlsmith [EMAIL PROTECTED] writes:
Why wouldn't you just use your existing Ethernet infrastructure putting
the IP phones inline between the wall jack and the PC? There are a
number of IP phones that have builtin switch/hub that allows the
On Tue, 20 Jan 2004, Ralph Blach waxed:
I would like to start using VOIP on Linux but I would like to start out
just using my sound card in my Linux box. Is there anyway to do this.
Yes, try using Asterisk:
http://www.asterisk.org/
Also, what service provider would I use to get my
On Tue, 20 Jan 2004, Ralph Blach waxed:
Can asteric be used with just a voice card. If so, how would I get
this going? Also, what carrier would I use connect to?
You'll want to do it on your Linux box.
Ie would would be my carrier.
Maybe these links would be your carrier:
On Wed, 21 Jan 2004, dkwok waxed:
What are the meaning of these Zap show channel output?
Caller ID string:
Owner: None
y
Real: None
?
Callwait: None
y
Threeway: None
y
Confno: -1
n
Propagated Conference: -1
n
Real in conference: 0
n
DSP: no
n
Relax DTMF: no
?
On Wed, 21 Jan 2004, [EMAIL PROTECTED] waxed:
Is there settings that can be adjusted in the Parked calls timeout before it hangs up
i want to try and hold the call for atlest 5-10 mins ..
but holds the call for about less then 1minute?
any suggestions or ideas?
Do you have an
On Wed, 21 Jan 2004, Steve Foy waxed:
Hi there, I'm having some trouble with getting Asterisk to make a call, I
think it should be quite easy, but anyway...
Using the following file contents:
##
Channel: Zap/3/TEL NUMBER HERE
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context:
On Wed, 21 Jan 2004, Bill Hamel waxed:
Hi,
Looking around I can't seem to find a way to show the number of agents currently
logged into a queue and if possible who they are. Is there a way to do this ?
Thanks
-b
I attached a patch I've been using to show the # of agents
(members) and
On Wed, 21 Jan 2004, Paul Mahler waxed:
I am trying to make outbound calls from my Asterisk client through a remote
Asterisk server with IAX.
In iax.conf on both sides
[dar]
context=trusted
secret=xx
type=friend
host=192.168.1.1
I'm not going to try and fix all of this, but if
PROTECTED]
Thanks again,
-bh
Quoting C. Maj [EMAIL PROTECTED]:
I attached a patch I've been using to show the # of agents
(members) and callers on a per queue basis. It adds a new
manager command, AgentQueues. It returns on the manager
interface the following for each queue
On Mon, 26 Jan 2004, Steve Foy waxed:
I'm just wondering about 'Zapateller'.
How exactly does it work!? I might be interested in employing it at work
here, but wondering if anyone's using it?
I think you can just put it in your dial plan:
exten = s,1,Answer
exten = s,2,Wait(2)
exten =
On Thu, 29 Jan 2004, Chris Hirsch waxed:
Currently a caller can press *3 to leave me a message in my own mailbox
on the FXS machine. Is there any way that I can make Asterisk monitor
the line after the machine has picked up and if it detects a *3 dump the
user into my Asterisk voice mail box
On Sat, 31 Jan 2004, Tomica Crnek waxed:
Hi everyone!
Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk
is connected to our national telecom company here in Croatia. When I call from
outside over this trunk to my company I get 'error in connection'
On Mon, 2 Feb 2004, Rick Smith waxed:
We have 1000's of Remote Call Forward #'s across the USA / Canada, which
forward into 1000's of 800 #'s in our call center.
Is it possible to automate a solution where Asterisk could dial a given
number, record the first 3 seconds of the call, save it
On Thu, 12 Feb 2004, [EMAIL PROTECTED] waxed:
My questions are as follows, (but before I begin; I know there is queueing
and some ACD functionality in *, but I need to do this externally. I want
the queueing decisions to be external because my central queue engine
handles things like email,
On Tue, 17 Feb 2004, Tomica Crnek waxed:
I have TE410P with two E1 links connected. It is working ok, but
suddenly, from time to time I got this and it goes on and on for a few
minutes during which period I can't establish new calls
== D-Channel on span 1 up
== D-Channel on span 1 up
On Mon, 16 Feb 2004, Jim Archer waxed:
First, can Asterisk be configured accept calls on a bunch of incoming
lines, answering with a greeting and telling the person that they will be
transferred to the next available operator. Then, can it watch all the
extensions, and route the calls to
On Wed, 18 Feb 2004, Jason Miller waxed:
Hello all I am new to this and was wondering if anyone could assist me or point me
in the right direction as to how to setup asterisk for a T-1 to be able to call a
PSTN number? I see plenty of walk throughs if you have analog cards installed but
On 28 May 2003, Steven Critchfield waxed:
8's
While I'm on the postgres bandwagon for now, I wouldn't want it in the
middle of a phone system doing heavy call loads either. Postgres also
has some downsides too. As I understand it, postgres doesn't understand
prepared statements, or at least
On Thu, 1 Jul 2004, Michael George waxed:
I want to have call forwarding (from the POTS) turned on at the close of work
and turned off automatically by *.
I can create a context that should do just that, but I need a way to have that
context spontaneously executed at a specific time.
I
On Fri, 2 Jul 2004, Brian D'Arcy waxed:
I'm using a TE410P, no irq sharing, and all extraneous devices disabled,
such as USB, Parallel etc. I'm getting a few IRQ misses according to
ZTTOOL.
8's
Can you try changing motherboards ? Just a guess, but it
seems like you've already made it quite
On Mon, 5 Jul 2004, Randy Bush waxed:
i am looking at iax to see if it is applicable to my needs. i
would appreciate any corrections of what i think i have understood
but probably have not.
Are we all supposed to guess what your needs are ?
iax uses udp and traverses nats. neither of
On Wed, 7 Jul 2004, lenz waxed:
Hello list,
I wonder if this is possible with Asterisk:
- While talking through Asterisk, I would like a client to start
recording a call by typing, say, #99#
I know it is possible to do it using an external monitoring application,
but I want to know
On Thu, 8 Jul 2004, Nauman Farooq waxed:
wondering if anybody knows this..does shady dial work only with a zap
interface or can it be configured to be used with SIP or IAX.
It is interface ambivalent.
--Chris
--
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide:
On Mon, 5 Jul 2004, Jeremy Kenney waxed:
Hello all I have a issue I am wondering if someone can help me
Here is my problem,
I have several queues setup for different numbers I want each queue to play
a custom message to the caller when calling in and then to the called
extention when the
On Thu, 8 Jul 2004, David Goldfein waxed:
Hello,
I am having an issue with making two simultaneous outbound calls.
When I dial, both phones try to take the same channel and it causes an
error. Anyone have any suggestions. My set up is as follows:
CO - PRI - ASTERISK - VODAVI(pbx).
On Fri, 9 Jul 2004, mattf waxed:
- Shady-dial (http://shadydial.sourceforge.net/)
Lead by some nice Europeans, they have a beta of it up and running
supposedly handling up to 10 agents per server, although I'm not sure of
exactly what level of 'Predictive' the dialer is(whether it
On Mon, 12 Jul 2004, Mamadou Lamine KA waxed:
Is there any alternative to Asterisk ZapBarge command for SIP and IAX channels?
Set up a silent meet me conference for the channels you want
to listen in on.
--Chris
--
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj
On Mon, 12 Jul 2004, Glen Hinkle waxed:
Are there any debugging tools for the digium zaptel cards that would
report the activity on the line, such as DTMF and/or connection
protocol?
* zttool is in the zaptel source directory
* you could issue set verbose 10 on the asterisk CLI
* you could
On Mon, 12 Jul 2004, Kyle Hagan waxed:
Can someone give me an example of what comes from the manager when a
call comes in on a PRI./T1 when a call comes in?
On the x100p i ger the following:
Event: Newchannel
Channel: Zap/1-1 - I need what would be populated in here
On Wed, 14 Jul 2004, Johannes van Hulst waxed:
8's
I called out on my X100T card.
It's not as reliable on an analog interface versus a digital
one, ie. a PRI. You will need to look at things like
callprogress in your config files and google for more.
--Chris
--
Chris Maj, Rochester
On Wed, 14 Jul 2004, Hall, Eric M. waxed:
I have been working on the music on hold part for a few hours today and
I found something that just doesn't sound right.
If I just run asterisk via service service asterisk start' everything
work but MOH
If I run it via asterisk -vgcd MOH
On Wed, 14 Jul 2004, [EMAIL PROTECTED] waxed:
On Wed, 14 Jul 2004, John Todd wrote:
This second
method also assumes that Dial is capable of intercepting DTMF, doing
some dialplan logic, and then re-connecting the two legs together
without hanging up either leg (see my many, many
On Wed, 14 Jul 2004, Kris Boutilier waxed:
Is it enough to simply plug an incoming T1 line in to a Digium T100p card or
should I pass the connection through some form of local CSU to provide
isolation, buffering, local diags and so on?
Plug it right in there.
--Chris
--
Chris Maj,
starts a service. Is this
something you added yourself ?
--Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Maj
Sent: Wednesday, July 14, 2004 5:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Music on hold
On Wed, 14 Jul 2004
=$?
;;
*)
echo Usage: asterisk {start|stop|status|restart|reload}
exit 1
esac
exit $RETVAL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Maj
Sent: Thursday, July 15, 2004 3:37 PM
To: [EMAIL PROTECTED]
Subject
On Thu, 15 Jul 2004, Celedonio Albarran waxed:
We have look at wiki and in the list archive and have not found any related
to it.
Have you looked at the consultants page on the wiki ? The
question you pose is fairly basic setup. It doesn't look
like you've even *tried* to do any
Hi James--
I got a dialer working without too many hiccups about two
months ago. It relies on changes to chan_agent, app_queue,
a PostgreSQL backend, a Tcl-* manager interface, a bunch of
Tcl glue, and some cron jobs. The results for each call are
logged in right through the phone key pad, and
On Mon, 5 Apr 2004, AstGrp waxed:
I am having an issue with Callerid (INBOUND). I have a system set up
with 4 companies sitting behind the system. On all of the companies
except of one of them, it displays callerid withh 'asterisk'. The other
company displays the callerid of the person
On Mon, 5 Apr 2004, William C. Ray waxed:
Hello i was wondering how i can change the IP address information for my Asterisk
box, IP addy, Gateway, DNS.
I have a smoothwall router that i am using and i am tring to put the Asterisk box on
the orange interface so if anyone can help me please
On Tue, 6 Apr 2004, Ryan Thrash waxed:
The number shows up, but I can't get the words to show on a local
bell line. The text always comes up as unavailable. In sip.conf for
each extension, I've tried:
callerid=VERTEX 2142618000
callerid=VERTEX 2142618000
Neither one works.
On Mon, 12 Apr 2004, Dragan Mickovic waxed:
Is it possible for asterisk to do an sql query in order to
get the member list of a call queue?
No, you will have to write code besides SQL in order to do
it. To go the C route, try modifiying app_queue. To use a
different language, you could code
On Tue, 13 Apr 2004, Dmitry Mishchenko waxed:
In other words can I receive information which we are usually getting in CDRs
during the time when the call is still active?
Yes, via the manager interface. Check manager.conf, it
lets * talk on port 5038.
--Chris
--
Chris Maj, Rochester
On Wed, 14 Apr 2004, Ezequiel Golub waxed:
Im trying to come up with a cost effective way to unite two PBX using VOIP.
My idea is that since most companys here (Argentina) are not ready cough up
the money to go to full-fledged VOIP, they might be willing to pay for a
hybrid-solution: a kind
On Wed, 21 Apr 2004, Laurent BURGY waxed:
Hi,
I am new in asterisk and i've bought a X100p and a TDM400...
First, you are probably eligible for support from digium
directly if you bought the hardware from them.
First of all, how can i verify my config files ?
You could try attaching them to
On Wed, 21 Apr 2004, Ben Merrills waxed:
Hi,
I have a couple of questions about MeetMe and call queues. I'm still
pretty new to Asterisk, but already having to write a Service Center
call manager for it (which I might add, our director has agreed to make
open source!).
That's great news.
On Fri, 30 Apr 2004, Dean Collins waxed:
Ian, I'd love to see an example of this.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain
Stevenson
Sent: Friday, 30 April 2004 1:47 AM
To: [EMAIL PROTECTED]
Subject: Re:
On Fri, 30 Apr 2004, Mark Elkins waxed:
Playing with time ranges - using the examples found in one of the
asterisk cook books... (pdf - page 17)
; After Hours
include = night_menu|00:00-08:00|Tue-Fri|*|*
include = night_menu|17:00-24:00|Mon-Thu|*|*
this gives...
... pbx.c:2962
On Mon, 3 May 2004, Joel Duffield waxed:
I am trying to get a way to have * forward calls that are dialed to an
extension, to end up at an extension on my old analog phone system.
I will have 7 lines coming into * using the new Digium cards via PSTN,
and then lines coming from * into the PSTN
On Sun, 16 May 2004, Bruno Fontana waxed:
I was trying to use TDMoE and I lasted with two problems. First of all I
can't configure the dynamic span to use CAS signalling but documentation
(by Mark) says that you can use any type of signalling (and this
includes CAS I guess).
Well just
On Mon, 17 May 2004, Robert Almeida waxed:
Could anyone tell me which is the recommended hardware to a system
running voicemail and conference, attending four E1 trunks and,
another, attending only one E1?
Can I use a PIII 850Mhz?
Maybe for a single port E1 card, maybe. You'll
On Wed, 19 May 2004, Mike Sturdee waxed:
Are there any variables or structure elements unique to a call that stay
till the end of a call -- including when caller enters a queue and then
bridged with agent. I am trying to get some variables about the caller in
I think the account code sticks
On Sun, 23 May 2004, William Zhang waxed:
Hi,
I have a TE410P with 3 E1 being enabled, some how it crashes for 2
times lately, I suspect it might be the channel setup issue, can
Does it crash immediately or after a fixed amount of time ?
anyone tell me if following part in zapata.conf is
Hank--
I was waiting for the 4th, 5th, or 6th email to reply...
BUT
Have you looked at the Wait(seconds) application ?
show application wait from the Asterisk CLI ?
Maybe try that before you issue an Answer() on the line ?
--Chris
On Mon, 24 May 2004, hank waxed:
- -
Don't judge me
On Wed, 26 May 2004, Fabio Donaggio waxed:
Hi to all!!
Here's my problem:
[cdr_pgsql.so] = (PostgreSQL CDR Backend)
== Parsing '/etc/asterisk/cdr_pgsql.conf': Found
May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql:
Unable to connect to database server localhost.
On Wed, 26 May 2004, Florent Guiliani waxed:
Hi all,
Is it possible and easy to make a CTI server with Asterisk?
Florent,
Yes, buy a computer and install Asterisk on it.
C = computer
T = Asterisk
I = install
--
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide:
On Wed, 26 May 2004, Maveric waxed:
I've noticed that when i pass a wait in an exten = that it doesn't allow
Are you talking about the Wait() application ?
'show application wait'
for dtmf tone input. Also on another note i've noticed that when using
Background() is what you want if you
On Thu, 6 Nov 2003, mattf waxed:
Sorry that got accidentally sent incompleted, here's the full post:
OK, here is the long drawn out description of how I am using Zap Barge and
Monitor:
8's
I record all outgoing calls using Kostya V. Ivanov's 'R'
patch to the Dial application, which was
On Fri, 12 Dec 2003, Derek Barber waxed:
but, if this is case then how can you run a call center with asterisk?
What if you have 40 simultaneous calls coming into the call center, most
calls would be missed, even if you have 40 available agents. Of course
one call should go to one agent,
On Wed, 17 Dec 2003, Anton Yurchenko waxed:
Hello,
I have a E1 with PRI signaling, is there a way to set that some of the
channels when the call is made from them to outside PSTN, will show one
callerID, and other group of channels will show other callerID.
Right now I have it like
On Wed, 17 Dec 2003, Patrick Cantwell waxed:
FYI: Doing an rm -rf zaptel asterisk in /usr/src, then doing a checkout from
12/08/2003, I no longer have this problem.. so it's something with the newer
code?
My nose is bleeding from CVS. Same thing with a
T400, had to comment out all fax
On Tue, 23 Dec 2003, Hubert Kiyimba waxed:
Dear members,
I am writing to inquire whether Asterisk can serve as video switching
software for the purposes of video conferencing over IP on a campus network.
Hubert
http://www.gnophone.com/
--
Chris Maj cmaj_hat_freedomcorpse_hot_info
On Sat, 10 Jan 2004, Lee Redmayne waxed:
Hi All
If I want to get my ADSI Phones (successfully connected off a Rhino Channel
Bank and TE410P) to connect to Asterisk to get their config downloaded, is
there something specific needed in extensions.conf for them to dial to get
this?
Thanks
On Sat, 10 Jan 2004, Steven Critchfield waxed:
On Sat, 2004-01-10 at 15:19, John Brown (CV) wrote:
busydetect=yes
callprogress=yes
musiconhold=default
signalling=pri_cpe
group=1
channel= 1-4
Well seems you haven't been on the list, or maybe you haven't been
paying attention since
On Sun, 11 Jan 2004, John Brown (CV) waxed:
THank you. Thats what I thought it should be.
Off to call the telco and tell them they are mucked up.
I'm wondering if I should do the same for my T400, as I seem
to be getting similar errors. Might not be just the telco.
I set one span to 1, to
On Mon, 12 Jan 2004, Andrew Thompson waxed:
I am curious. I understand that features can be pushed to an ADSI phone that
make navigating your own voicemail easier, and for other internal things.
But, does anyone push this data outside of their own phone network?
Example: I am at home with
On Mon, 12 Jan 2004, marin blu waxed:
I'm trying to install * on Mandrake 9.2/P4, but under asterisk - make clean;make
install there is the following error:
How about:
make
then:
make install
--
Chris Maj cmaj_hat_freedomcorpse_hot_info
Pronunciation Guide: Maj == May
On Tue, 13 Jan 2004, Lane Hoskins waxed:
We have 8 lines coming into an ADTRAN channelbank that then goes to the
* server via a T100P card. I need to route lines 1 and 2 to everyone
when a call comes in on either of them. I also need lines 3 - 8 to ring
first at specific sip extensions
On Tue, 13 Jan 2004, Ted Cabeen waxed:
Martin Pycko [EMAIL PROTECTED] writes:
sure, use the 'n' option of the queue and put voicemail app as the next
priority
Will that work? From my read of the code, the timeout parameter is
only checked while the call is being sent to an agent's
On Wed, 14 Jan 2004, Gary Franczyk waxed:
I'd like to configure a voice recording system using Asterisk and a
Tormenta2 Quad T1 card. A co-worker was able to create this system a while
back with Bayonne and a Dialogic card, but I would like to do the same thing
with much cheaper hardware. I
On Wed, 14 Jan 2004, calvis waxed:
Thanks for the link.
This is an interesting article on Asterisk. I was hoping to send him some
kudos, but his website isn't working at http://www.bschwarz.com/. And I
just noticed the guy lives near me!
Does anyone know if he hangs out on the list?
On Wed, 14 Jan 2004, [EMAIL PROTECTED] waxed:
On Wed, Jan 14, 2004 at 07:34:20AM -0500, Troy Settle wrote:
I'm not completely conversant on how GPL software can be committed to
the kernel, but I believe it can be done under the contrib/ directory.
I think the *BSD kernel device
On 8/17/23 05:04, Federico wrote:
Yes that are, but how do I use them to execute a part of the dialplan, once,
when Asterisk starts up.
Systemd provides "ExecStartPost=" option to run more commands eg.
cp /usr/local/src/asterisk/sample.call /var/spool/asterisk/outgoing/
--
On 8/17/23 12:44, John Harragin wrote:
You should be able to define multiple data sources. However I'm having my
own issues. I have my dialplan accessing one maria database which is hosted
locally on the asterisk server then logging cdr with odbc adaptive which
connects to maria on a remote
On 8/21/23 08:23, Jerry Geis wrote:
I am using asterisk 18.14.0 and chan_sip.
confbridge has dsp_drop_silence=yes
The conf joins all the endpoints in a one-way conf.
60+ devices and packets choppy or dropping audio.
The CPU is decent at Intel(R) Xeon(R) CPU E3-1240 v5 @ 3.50GHz
What else
risk Forums are a great place to post these kinds of
questions in the future: https://community.asterisk.org
Regards,
--
鸞 C. Maj, TechnoCaptain
Penguin PBX Solutions
Denver 720-32-42-72-9
Beyond 1-833-PNGN-PBX
http://PeN
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