We have switchboard application ( PC+browser+Java ) with quite a rich
feature set.
It talks to * via manager port.
Works as a call center too.
However, it is not open source.
If you are interested in, please contact me directly.
Best regards Pertti
Keith D'Atrio wrote:
Hello All
I am
Thanks for all the replies.
Can someone tell me if it is possible to put two of
these X100P cards into the same machine to order to gain access to two BT
landlines ?
Would it also be possible for someone to outline in
a bit more detail the procdue for limiting which phones have access via
--On Saturday, April 10, 2004 10:42:26 +0100 Paul Tyreman
[EMAIL PROTECTED] wrote:
Thanks for all the replies.
Can someone tell me if it is possible to put two of these X100P cards
into the same machine to order to gain access to two BT landlines ?
I believe so although problems have been
Paul Tyreman wrote:
Thanks for all the replies.
Can someone tell me if it is possible to put two of these X100P cards
into the same machine to order to gain access to two BT landlines ?
Would it also be possible for someone to outline in a bit more detail
the procdue for limiting which
What I want to do is have the asterisk server sat
in my house and used by my family to access the BT landline and to recieve calls
made to that landline. If it is not possible to do the auto attendant
thing then so be it, I will just have all phones in my house ring when a call is
made on
Hi!
I'm really hope you can help me solve a little mystery, the mystery is
probably just my misunderstanding ! sorry...
I've got an iaxy talking to my * box which connects to two providers.
I'm running the stable release of the pbx.
The only thing is that when dialling from the iaxy the ringing
Being bored to death by these long weekends with nothing to do?
Why not go bounty-hunting?
There are some feature requests in the bug tracker with monetary bounties attached.
* Windows manager
* FreeBSD Zaptel drivers
http://bugs.digium.com/bug_view_page.php?bug_id=847
* IAX
What version of the Asterisk code are you running? 1_0 stable is definitely
broken wrt ringback, and the latest stuff seems really broken in all kinds
of ways. After seeing that others were having similar problems, and that
someone had solved many of them by rolling back to the CVS version from
Brian,
I need to roll back to an earlier version to identify a different problem,
but I dont have the cvs checkout command string that includes a date. Can
you post how to do that please?
Rich
What version of the Asterisk code are you running? 1_0 stable is definitely
This perhaps is a newbie question or I have been up too late working on
this. Shouldn't I be able to dial internal extensions via the
inboundanalog1 menu? When I dial an extension from an external call
to the inboundanalog1 menu, I get a busy and a hangup?
Any suggestions?
[extensions]
exten =
Sure. I used this to get the 3/5 version:
cvs co -D 20040305 zaptel asterisk
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Saturday, April 10, 2004 9:13 AM
To: [EMAIL PROTECTED]
Subject: RE:
Kevin wrote:
exten = 0,1,Dial,${P6601}
what does ${P6601} contain?
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your
${P6601}=SIP/P6601 and is a SIP Cisco Extension. The internal
extensions dial properly and the SIP phone is properly associated with
the extensions context. I want to be able to dial internal extensions
from the incoming analog line so I included the extensions context in
the incominganalog1
I am terribly sorry to bother the list with such generic and bizarre
problems, but I have been racking my brain with these for the last week
working on it for at least 60 hours. If anyone can even point me in the
right direction I would be eternally grateful. So without further adu
here are my
Pertti Pikkarainen [EMAIL PROTECTED] wrote:
We have switchboard application ( PC+browser+Java ) with quite a rich
feature set. It talks to * via manager port.
Works as a call center too.
However, it is not open source.
Why not?
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/
Just when I thought I couldn't be wrong, I was wrong. We have
woodpeckers that drill into the arial telephone cables, and water seeps
through the holes and partially grounds the tip and/or ring wires
causing hum. I thought the hum/buz on my lines was a telco problem.
The Qwest HQ noise team
When I reboot the computer running * and run * via asterisk -gc
everything starts without any warnings, notices, or errors. At that
point none of my SIP clients login to *. If I do a sip debug it doesn't
even show the clients trying to connect, however on the X-Lite logs it
is sending
Test.
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Keywords
T1 Q.931 isdn disconnect cause codes itu standard libpri
Dont know if anyone wondered what q.931 cause codes are
but i wishwe could get these back into the dial plan as a var
Standard Q931 Codes
Decimal Value Hexadecimal Value
Definition
1 01 Unallocated (unassigned) number.
This
I wondered if anyone had had any success in receiving faxes with Asterisk
and if so if they might point me in the right direction ?
Chris
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To
It sounds like a layer three problem. Are all of the Subnet masks and
gateways configured correctly? Obviously, you can ping, but you are
unable to connect without the ping. It sounds like both the SIP phones
and the IAX customers are unable to find a route back to your asterisk
server, but
Mike,
The Qwest HQ noise team assures me that my lines are within spec. Sure
enough, when I listen on the test set the lines are clear.
The lines terminate at an Adtran 750 channel bank on my * system. When
I reconnect the lines to the channel bank and make a call, I get the
hum/buz
I wondered if anyone had had any success in receiving faxes with Asterisk
and if so if they might point me in the right direction ?
There's a good pointer to the Opencall.org SoftFax solution at
http://www.voip-info.org/wiki-Asterisk+fax that I was able to get working
on the stable CVS tree..
Paul Tyreman wrote:
What I want to do is have the asterisk server sat in my house and used
by my family to access the BT landline and to recieve calls made to
that landline. If it is not possible to do the auto attendant thing
then so be it, I will just have all phones in my house ring when a
Sure you need to add -r v1-0_stable in the checkout command
see
http://www.asterisk.org/index.php?menu=download
All the best, Chris
On Sat, 10 Apr 2004, Paul Tyreman wrote:
Hi,
I downloaded Asterisk using this command a couple of weeks ago...
# cd /usr/src
# export
When we were first installing these SIP phones, the time and date was
displayed on the top status line. Firmware version 6.3. Using dhcp, tftp,
http, and ntp servers for configuration all on the LAN (* box).
Now that we have these working with *, the time/date display is missing.
Any ideas?
--On Saturday, April 10, 2004 11:55:26 +0100 Paul Tyreman
[EMAIL PROTECTED] wrote:
What I want to do is have the asterisk server sat in my house and used by
my family to access the BT landline and to recieve calls made to that
landline. If it is not possible to do the auto attendant thing
Rich Hi,
After Qwest pronounced my circuits as within spec. (yes, disconnected
from the house) I listened on the lines with my butt set. Clear of all
noise and hum. I then got my box of Cat3 and laid a circuit around the
outside of the house and into my lab. Still clear of all noise and
To the list ...
The problem appears to be fixed. Short answer: interrupts.
When looking at cat /proc/interrupts
it was seen that the wct1xxp card was sharing interrupts
with a.o. the eth0 main ethernet driver.
The solution we simple: move the T1 card to another PCI
slot.
Upon checking cat
Hi again,
Just wanted to add to my previous post that you should run the modified
command to download the stable version ie
cvs checkout -r v1-0_stable asterisk
in another directory from /usr/src (it will create 'asterisk' directory
for you)
or first use 'mv' on your existing asterisk...
Ignoring the phones, can you reach the ntp server from Linux? I've had
trouble with ntp servers being unresponsive.
There is a very useful Windows utility called AboutTime by Paul Lutus
http://www.arachnoid.com/abouttime/index.html. I use this to first
check the time servers.
Mike
Tom
Hi,
Glad that your problem was solved, but we are still exerpiencing a
similar problem but our interrupts show:
0: 162034 XT-PIC timer
1:234 XT-PIC keyboard
2: 0 XT-PIC cascade
5:782 XT-PIC eth1
7: 7448
Sorry to sound stupid, but where can I get copied
of the Asterisk manual ?
What is the VoIP wiki and where can I get that too
?
Thanks, Paul.
-Original Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain
StevensonPosted At: 10 April 2004 17:20Posted To:
The instructions are already posted on Asterisk's website:
http://www.asterisk.org/index.php?menu=download
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Tyreman
Sent: Saturday, April 10, 2004 11:12 AM
To: [EMAIL PROTECTED]
Subject:
Mike,
After Qwest pronounced my circuits as within spec. (yes, disconnected
from the house) I listened on the lines with my butt set. Clear of all
noise and hum. I then got my box of Cat3 and laid a circuit around the
outside of the house and into my lab. Still clear of all noise and
Tan,
Scary ...
What we used to see was quite few (and sporadic) notices and
warnings in the /var/log/messages file reporting PRI
trouble. Especially event 6 and event 8 if I recall. These
notices and warnings have disappeared since we resolved the
interrupt issue. Because the implementation is
Brian Cuthie wrote:
What version of the Asterisk code are you running? 1_0 stable is definitely
broken wrt ringback, and the latest stuff seems really broken in all kinds
of ways. After seeing that others were having similar problems, and that
someone had solved many of them by rolling back to
--On Saturday, April 10, 2004 17:47:24 +0100 Paul Tyreman
[EMAIL PROTECTED] wrote:
Sorry to sound stupid, but where can I get copied of the Asterisk manual
?
http://www.asterisk.org/index.php?menu=support#handbook_project
What is the VoIP wiki and where can I get that too ?
The wiki is a
Paul,
http://www.voip-info.org/wiki-Asterisk
Wim
- Original Message -
From:
Paul
Tyreman
To: [EMAIL PROTECTED]
Sent: Saturday, April 10, 2004 6:47
PM
Subject: Re: [Asterisk-Users] Re:
Analogue telephone cards for the UK
Sorry to sound stupid, but
Probably for the same reason you charge for your services. Software takes
time and skill to write. And while I'm grateful that people like Mark
release their apps to us as open source or under GPL, I don't begrudge
anyone from wanting to actually make a living.
-brian
-Original
Here is what we get:
Apr 10 18:10:34 WARNING[-1179604048]: chan_zap.c:6026 zt_pri_error:
PRI: Read on 24 failed: Unknown error 500
Apr 10 18:10:34 NOTICE[-1179604048]: chan_zap.c:6740 pri_dchannel: PRI
got event: 8 on span 1
We were getting around 5 messages per second. I turned off the usb
When I installed 1_0_STABLE, ringback stopped working completely on all
calls through the TDM400P. I can't recall if the SIP phones stopped
generating ringback also.
Latest builds (as of yesterday) seem to have problems with dropouts,
especially with IAX connections. I was seeing dropouts and
Hi
everyone,
Is it possible and
how to add one or more digits as prefix to all calling numbers coming from one
direction. What I want to do is to add 0 to calling numbers for all calls coming
from one E1 span to my Asterisk box. So, if the call is coming from number
1234567 it will be
Tan,
My warnings notices have stopped completely.
I was getting them once every few hours though, instead of
once every few minutes.
However, in the process of trying to resolve issues, we also
rolled back the codebase to 3/5/2004. This *maybe* a red
herring, but I see several other people on
On Sat, 2004-04-10 at 20:24, Tomica Crnek wrote:
Hi everyone,
Is it possible and how to add one or more digits as prefix to all
calling numbers coming from one direction. What I want to do is to add
0 to calling numbers for all calls coming from one E1 span to my
Asterisk box. So, if the
On Sat, 10 Apr 2004, Uriel Carrasquilla wrote:
After rebooting my asteriks server, e-mail notifications are no longer being
sent after a voice-mail is left.
I can see the messages in /var/spool/asterisk/vm.
has anybody had the same experience? how was it resolved?
Perhaps your MTA daemon
To whom it may
concern,
When dialing out an
800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound. This
is not due to a problem on your Asterisk or your line- the bad soundeffect
occurs in VoicePulse. (just spend lots of time finding that
out)
Assaf
BenharooshMCP,
Sending e-mail on my server has never worked.
It always bounces back to the server saying it was undeliverable !?
Its a right pain.
-Original Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathaniel
PowningPosted At: 10 April 2004 19:54Posted To:
Does anyone know if you can put two of the X100P cards in
to the same machine and have access to two landlines ?
I just need to know if it's worth buying two or not
!
Thanks, Paul.
On 2004 Apr 09, at 10:09, Ed Rubright wrote:
Did you have to apply a patch to get this to work, or is it in CVS?
Bug # 413, among others.
Please don't top-post.
-Tilghman
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On Sat, 2004-04-10 at 20:01 +0100, Paul Tyreman wrote:
Does anyone know if you can put two of the X100P cards in to the same
machine and have access to two landlines ?
I just need to know if it's worth buying two or not !
Thanks, Paul.
Yes. You can put in as many cards as you have open
Hi all,
If I'm just using Asterisk as PBX and calls going
through between ouside lines and inside extensions,
(not using any softphone running on the asterisk pc),
does what soundcard I use affect voice quality at all?
Do I have to get a full duplex soundcard?
Thanks
Ron
Paul Tyreman wrote:
Does anyone know if you can put two of the X100P cards in to the same
machine and have access to two landlines ?
I just need to know if it's worth buying two or not !
Thanks, Paul.
I run 4 X100P's in our asterisk box. Just make sure you give each card
it's own IRQ.
Hello,
I am very new to asterisk and voip in generaland so far have managed
to get the FXO card and a few sip phones working fine. My question is where does
the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e
FXS) ? Or is it to be used as a line (i.e FXO)? Or it can
Hello,
The Sipura is used as an FXS adapter, in that it
allows you to plug a phone into either of it's 2 lines and have a connection to
asterisk. I'm sorry to tell you that it can't be used as an FXO
adapter.
- Joshua Colp.
- Original Message -
From:
San Singhania
San Singhania wrote:
Hello,
I am very new to asterisk and voip in general and so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e
Hi!
I just wondered if anyone would mine posting their successful jitter
buffer settings here for me if they get a moment ??
I've spent a few hours trying to set the jitter buffer up reasonably
logically and can definitely tell it makes a difference and can introduce
latency and echo if setup
I purchased the DigitNetworks VoIP Starter Kit Full
(FXO Card GrandStream BudgeTone-100 IP Phone)
To tell the truth, I can't believe I've got it
workingthis far! Most everything is working.
However, I'm having a few problems outlined
below:
Using XLite: - Working inside the LAN
Ican
Steven Critchfield wrote:
On Sat, 2004-04-10 at 11:51, Bob Klepfer wrote:
(I *have* noticed RAM almost completely filled, but no swap used...a
reboot freed a bunch and I think that fixed some issues. We're a small
company and restarting * or rebooting the server isn't that big a deal.)
Hi!
Could someone please tell me where in the CVS tree the IAXY firmware is
and the utility used to install/flash the IAXY?
Or a perhaps cvs command that shows me which directories are available in
the CVS?
I did 'man cvs' but can't find such a command or flag.
Also is there any documentation
Sorry- wrong
observation. The problem is when placing a call to IAX from a Cisco
7940.
To whom it may
concern,
When dialing out
an 800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound.
This is not due to a problem on your Asterisk or your line- the bad
soundeffect
On Sat, 2004-04-10 at 17:51, Bob Klepfer wrote:
Steven Critchfield wrote:
On Sat, 2004-04-10 at 11:51, Bob Klepfer wrote:
(I *have* noticed RAM almost completely filled, but no swap used...a
reboot freed a bunch and I think that fixed some issues. We're a small
company and
At 12:58 PM 4/5/2004 -0500, Steven Sokol wrote:
I regret that I've only used MeetMe a few times, and only up to two users.
Perhaps others that are using MeetMe could comment on the number of
concurrent conferences and total users they have asterisk running with.
The
specs of the systems
On Sat, 10 Apr 2004, Robert Jackson wrote:
Thanks for the info. I am not sure how to disable iptables, but I will
be scouring the net for the next couple of hours or so. I simply
couldn't believe that * was as unstable as it has been seeming. At
least now I know that I'm not crazy. Rich,
Lots of responses, but here is my 2 Cents of what
works well for me
Downloaded my local NPA/NXX combinations from
following web site.
http://members.dandy.net/~czg/search.html
Added them to the internal Asterisk DB with DB put
So when an outbound call comes, I strip off the
NPA, NXX, do a
Hello-
I have several Sipura SPA-2000's at different locations (all behind Linksys WRT54G
boxes). When setting:
nat=yes
qualify=yes
Things work properly about 90% of the time, however, if a remote end loses the
connection briefly, then asterisk can't see the adapter until the next
Well, the stupidity just keeps on coming. Thanks for all of your posts
I followed all of your advice and was able to resolve the problem. As I
wrote in the first post I have been banging my head against this problem
for around 60 hours, until I finally gave up and decided to post.
The answer
A somewhat plussed asterisk admin wrote:
The answer is a bit embarrassing as I should have checked this issue
within minutes if not hours of these problems cropping up.
Just for the record--I've been in this boat my fair share, too--this is
why some of the old hands on the list are a little
On Fri, 8 Apr 2004, Hermann Wecke wrote:
I'm trying to add 2 FXO cards
ztcfg -vv is reporting only 1 card:
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
Please post your zaptel.conf file, as well as the
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