That did it.
thank you, thank you, thank you,
-reed
Marc Sutter wrote:
Hi,
had the same problem... and we wrote a patch.
This patch's are for speech_tools 1.2.3 and festival 1.4.3.
to use in the corresponding directory with:
#patch -p1 patch..
Hope this help. If so let it know.
Have
Hi,
I am a newbie in asterisk, i could compile it and run it with no problem on
a RedHat 9. In the same box, i got a linejack and a phonejack cards and i
downloaded the CVS driver from quicknet. This 2 card were working in a
openh323 (openphone and pstn) project with gnugk on a RedHat 9.
I am
I'm a newbie too -- search the archives for ztdummy.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Raul
Elizondo (wizardteam)
Sent: Sunday, May 02, 2004 2:32 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] phonejack and linejack in the same system
On Sunday 02 May 2004 12:30, Kevin Walsh wrote:
I'm not sure if software support for the line reversal monitoring is
desirable in the driver.
Probably not - but surely isn't is a well-accepted method that software is
often used to fix bad hardware design?
I firmly believe that half the
Hi,
I was searching along the net for a tftp server
nat-friendly, in order to provide new firmware
to our budgetones, which are 90% of time nat'ed.
I came across this one:
http://troja.ath.cx/~zond/jtftp/
works ok. is written is java, under linux.
I'm using it now and seems ok. very simple,
I personally download the TFTP server provided free by SolarWinds.
I simply open up the required port and is very happy.
http://support.solarwinds.net/updates/New-customerFree.cfm
Regards,
Ian.
---
Outgoing mail is scanned but not certified Virus Free.
Checked by AVG anti-virus system
sure... but the one I suggested is meant to be run
as a service on a linux server, maybe the same
as your asterisk box.
why having a winzoz machine to provide only tftp service ?
:)
Matteo.
Il dom, 2004-05-02 alle 14:05, Ian Pilkington ha scritto:
I personally download the TFTP server provided
Is anyone on the list successfully using iconnecthere
behind NAT?
Yes (unless it broke during the last 12 hours).
I gave my daughter a SIP phone and an Iconnecthere account.
I have successfully used Sipura 2000, a Grandstream and a
Cisco ATA box with that account, and yes, she is behind a
I'm trying to get Asterisk to talk SIP to Vocal and so far have only
managed to get it partially working. Calls in from Vocal are working
fine but outbound calls aren't.
In sip.conf I have:
[ivv]
secret=SECRET
username=08452416761
host=sip.intervivo.net
Hello,
I'm using an AGI program written in C to manage incoming calls to some
extensions. Its being used for a small call center (20 people).
When the call comes in, the caller can listen the directory menu and
then dial the extension. The AGI program is called and get one of the
available
Hi
Let's say i have a call to a extension 115.
But i'm under the extension 118 how take the call from 115 to my extension using * ??
Thanks alot.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening:
localhost*CLI -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack --
On Saturday 01 May 2004 18:00, Hermann Wecke wrote:
On Fri, 30 Apr 2004, Mark Elkins wrote:
Looking at pbx.c - I'm not sure if I should change the end time
(ie midnight) to either 23:59 -or- 00:00.
it is 23:59
23:59 will work - but what happens to calls then between 23:59
and midnight?
J Poz wrote:
-- SIP/jtest-6a1e is circuit-busy
== Everyone is busy at this time
May 2 11:47:58 WARNING[1226062640]: pbx.c:1198 pbx_extension_helper:
No applica
tion 'DialCongestion' for extension (sip, 22, 2)
Come on... Asterisk is telling you EXACTLY whats wrong... SIP/jtest is
On Sun, 2004-05-02 at 13:48 +0200, Brancaleoni Matteo wrote:
Hi,
I was searching along the net for a tftp server
nat-friendly, in order to provide new firmware
to our budgetones, which are 90% of time nat'ed.
I came across this one:
http://troja.ath.cx/~zond/jtftp/
I found gs_config
Jeremy,
I'm missing something hereI understand the congestion part - no problem, I can take that out of extensions.conf..
But I need help in the real problem which is the Auto-congesting and circuit-busy part. Why is it saying the circuit is busy? If siphone "jay" dials "jaytest" it says
Hi!
I'm trying to get Asterisk to talk SIP to Vocal and so far have only
managed to get it partially working. Calls in from Vocal are working
fine but outbound calls aren't.
I haven't looked at your settings, but two days ago I upgraded to latest
CVS and since then I am unable to place
On 1-May-04, at 4:51 PM, Jeb Campbell wrote:
On May 1, 2004, at 2:04 PM, Ryan Courtnage wrote:
I'm trying to get fax detection to work.
Hi Ryan, in stable FAX_DETECT is turned off by default in the code
(dsp.c).
I'm personally using it with spandsp and having no problems, but YMMV.
If you
Philipp von Klitzing wrote:
Hi!
I'm trying to get Asterisk to talk SIP to Vocal and so far have only
managed to get it partially working. Calls in from Vocal are working
fine but outbound calls aren't.
I haven't looked at your settings, but two days ago I upgraded to latest
CVS and
On Sat, 1 May 2004, Dean Collins wrote:
Yes but no information about how this will operate, what regulation or
restrictions on joining, what connection protocols will be used etc etc
agreed - you see alot of business fluff - but the technicals are very
important and on many of these ventures
On Sat, 1 May 2004, Rich Adamson wrote:
now. But if you have a look at this page -
http://www.freeworlddialup.com/advanced/iax you will find that you can now
use FWD with IAX2 along with SIP :)
FWIW, I just moved our FWD account to iax2, and it works rather well
with *. The referenced
On May 2, 2004, at 1:02 PM, Ryan Courtnage wrote:
I'm personally using it with spandsp and having no problems, but YMMV.
If you want to enable it, goto line 60 of dsp.c and uncomment that
#define so it looks like this:
/* Define if you want the fax detector -- NOT RECOMMENDED IN -STABLE
*/
Steve Underwood [EMAIL PROTECTED] wrote:
Wake up.
Sorry, I must have drifted off for a while. Thanks for the alarm
call.
The reversal detection is a complete waste of time. Totally unnecessary.
Pointless. A line break detector would have much more use, as it would
give a reliable
If you are indeed running RH9.0 you don't need to install any extra ixj
drivers.
This distro has all drivers included and they work fine.
ztdummy has nothing to do with chan_phone.
However, you do need to create the device nodes and load the driver with
modprobe.
It wasn't clear to me from the
Joe,
Inline comments...
now. But if you have a look at this page -
http://www.freeworlddialup.com/advanced/iax you will find that you can now
use FWD with IAX2 along with SIP :)
FWIW, I just moved our FWD account to iax2, and it works rather well
with *. The referenced web page
On Saturday 01 May 2004 16:42, Gavin Hamill wrote:
PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE can some Kind and Worthy
soul spend a little time in getting this really important feature
implemented? You would have the undying gratitude of thousands of X100P
users all round the world! :D
I've come late into this thread, so I risk saying things that you all
will just shake your heads at and say, Duh!
Historically, though, from WAY back in the days of electromechanical
switches, reverse battery mainly provided answer supervision. Its
usefulness pretty much went away with the
Who ever works with chan_vpd please look over
this:
http://bugs.digium.com/bug_view_page.php?bug_id=961
Thanks,Brian
On Sunday 02 May 2004 19:33, Jon Lawrence wrote:
I emailed sales at digium asking whether the new module supported
international (ie non bellcore) cli. The answer was yes, but it's not yet
implemented in the driver - driver implementation is in the pipeline
apparently.
Whether this means
WARNING!!! WARNING!!! WARNING!!!
If support for UK callerid is a MUST then get off your butt and raise some
hell. I sit here and read thru these emails where people chatter back and
forth about it. I don't see anyone trying to drum up intrest in this! If
this is something you want MAKE IT
On Sunday 02 May 2004 21:16, brian k. west wrote:
I'm not saying code it, I'm not saying PAY for it. I'm saying find someone
that can make it happen and ask for help.
Believe me I'm already doing this - I've rallied the local Linux geek
community as much as I can and there are a couple of
I can connect to FWD on my asterisk - but FWD only see me as an external
SIP agent and not a SIP client of the FWD network. DOn't know exactly why
- so would luv to compare your conf files.
There is really not much you can do about that with your local
configuration.
The problem is on
Hi,
I'm thinking about getting a couple of Cisco 12SP+
phones to use on my Asterisk system.
I have just bought a Cisco 7960, and they are
great, but too expensive to buy a lot of them, so I though I might try the 12SP+
ones.
I have seen in the archives that the phones work on
Asterisk,
Jon Lawrence [EMAIL PROTECTED] wrote:
I emailed sales at digium asking whether the new module supported
international (ie non bellcore) cli. The answer was yes, but it's not yet
implemented in the driver - driver implementation is in the pipeline
apparently. Whether this means that the
I've recently begun integrating an Asterisk system into my house. I
purchased the Dev Kit a year or two ago when Digium was selling it as a
Z-plex 10 channel bank with the T100P.
I've recently found that when I connect the serial monitoring port to my
system it introduces a noticable hum on my
brian k. west [EMAIL PROTECTED] wrote:
If support for UK callerid is a MUST then get off your butt and raise some
hell. I sit here and read thru these emails where people chatter back and
forth about it. I don't see anyone trying to drum up intrest in this! If
this is something you want
On Sun, 2004-05-02 at 22:07, Kevin Walsh wrote:
Jon Lawrence [EMAIL PROTECTED] wrote:
I emailed sales at digium asking whether the new module supported
international (ie non bellcore) cli. The answer was yes, ...
The Digium shop (http://store.yahoo.com/asteriskpbx/newitd1pofxo.html)
says
What does it mean ?
May 2 20:37:21 WARNING[1205250992]:
chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16386,
dropping
Asterisk CVS-05/02/04-23:04:14 built by [EMAIL PROTECTED] on a i686 running Linux
from
cvs checkout -r v1-0_stable asterisk
I think this was fixed in CVS-HEAD because I do not
see that message in the src at all while looking to see if t was
fixed.
bkw
- Original Message -
From:
Serge Oleinikov
To: [EMAIL PROTECTED]
Sent: Sunday, May 02, 2004 2:40 PM
Subject: [Asterisk-Users] IAX2
On Sun, May 02, 2004 at 09:07:37PM +0100, Kevin Walsh wrote:
The same Digium shop page suggests that two PCI slots would be required.
I'll assume the card is too fat, with the daughter board(s) fitted, to
fit into a single slot.
This is something I would like to see confirmed, does this card
The same Digium shop page suggests that two PCI slots would be required.
I'll assume the card is too fat, with the daughter board(s) fitted, to
fit into a single slot.
This is something I would like to see confirmed, does this card really
take 2 pci slots? I had hoped to make use of
hi everybody,
just upgraded my bri-stuff driver to 0.0.2rc20a. now i have a strange
problem :-(
i have immediate = no but when i pickup the phone i get :
*CLI == D-Channel on span 1 up
-- Extension 's' in context 'default' from '6294094' does not exist.
Rejecting call on channel 2, span 1
Need some help with modules.conf, and basic RH9 linux skills.
I've installed the new TDM04B 4-port FXO card and its working. After
a reboot, when I do lsmod I see the wcfxo module but not the wcfxs
even though both are listed modules.conf.
If I modprobe wcfxs, then lsmod has both modules
Hi.
flame mode on
Need some help with modules.conf, and basic RH9 linux skills.
perhaps wrong list? see linux kernel howto...
I've installed the new TDM04B 4-port FXO card and its working. After
a reboot, when I do lsmod I see the wcfxo module but not the wcfxs
even though both are listed
Mark J Elkins wrote
Um - Digium wants you to buy their hardware - but there is a CLID
issue.. would it not make more financial sense to insert a dumb ISDN
card (or two), and upgrade your PSTN to ISDN??? Would this not assist
Digium in making sure CLID worked in the UK???
Isn't this a bit like
I'm using the stable
branch. Is voicemail or voicemail2 deprecated?
TKS
Paul
[EMAIL PROTECTED]
Scrive Paul Mahler [EMAIL PROTECTED]:
I'm using the stable branch. Is voicemail or voicemail2 deprecated?
RGH!!!
ages passed when voicemail was sent to /dev/null and voicemail2
moved to voicemail...
current voicemail is the old voicemail2 voicemail doesn't
exist any more.
I've installed the new TDM04B 4-port FXO card and its working. After
a reboot, when I do lsmod I see the wcfxo module but not the wcfxs
even though both are listed modules.conf.
If I modprobe wcfxs, then lsmod has both modules showing.
why you need wcfxs on a quad-fxo ?
Because
immediate=no is in the right position into zapata.conf?
ie before the channel=XX you're picking up?
matteo.
Scrive FastJack [EMAIL PROTECTED]:
hi everybody,
just upgraded my bri-stuff driver to 0.0.2rc20a. now i have a strange
problem :-(
i have immediate = no but when i pickup the
On Mon, 2004-05-03 at 00:11, David J Carter wrote:
Mark J Elkins wrote
Um - Digium wants you to buy their hardware - but there is a CLID
issue.. would it not make more financial sense to insert a dumb ISDN
card (or two), and upgrade your PSTN to ISDN??? Would this not assist
Digium in
Hola,
if you have overlapdial=no in zapata.conf then * will jump into the
s extension on a NT span (this way you can use DigitTimeOut and
ResponseTimeOut to make patterns like _X. work as expected.).
So, either you create an s extension, e.g.:
exten = s,1,DigitTimeOut(3)
or you set
I am using the following macro to dial a ZAP channel. When I dial in, *
answers and I go to voicemail. I never hear any ringing, though. It doesn't
work with the Ringing command before or after the Dial command.
[macro-zapdial]
;
; call a ZAP extension for ${ARG2} seconds, and then voice mail
;
(We prefer plain text posting over HTML posting) ;-)
Let's say i have a call to a extension 115. But i'm
under the extension 118 how take the call from 115
to my extension using * ??
This has been covered a number of times on the list and in the wiki but it's
such a nice day outside I can't be
The reversal detection is a complete waste of time. Totally
unnecessary. Pointless. A line break detector would have much
more use, as it would give a reliable disconnect detection on
many lines.
Hmm.. but wouldn't reversal detection help those people who are in the UK
and want to receive
I got it! Nothing like posting to the mailing list when you're going to look
stupid to help you find the answer yourself!
The answer is to use waitforring(1)!
Thanks!
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
On Monday 03 May 2004 00:02, Paul Crick wrote:
Hmm.. but wouldn't reversal detection help those people who are in the UK
and want to receive caller ID?
Yes and no. If we could leave the FSK modem running all the time, then
hardware support for the polarity switch wouldn't be as vital.
On Monday 03 May 2004 00:02, Paul Crick wrote:
Hmm.. reading up, you could probably get away with just monitoring for the
alert tone then?
And this time round, I actually read your message.
Yes, you're quite right - it might only be a tiny 'bleep' on the line, but if
the line is currently
Can anyone help. I've changed the extensions.conf file as follows:
extensions.conf
[sip] ; context for X-Lite Clientsexten =11,1,Dial(SIP/jay,20,tr)exten =22,1,Dial(SIP/jtest,20,tr)
I'm still getting the Auto-congesting error (and circuit-busy). Does anyone know what is causing this in such a
Has anyone tried a Vina T-1 Integrator as a channel bank with Asterisk?
They appear to be plentiful, but I want to make sure I'm not buying a
brick.
THX/BDH
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Simple solution on redhat machines
In the zaptel source tree (At least the CVS one) there is a file called
zaptel.init. This is a script that will allow you to install all needed
modules. To use it do this:
cd /usr/src/zaptel
cp zaptel.init /etc/init.d/zaptel
chkconfig --add zaptel
chkconfig
I just got the X100P and the TDM400P with one module on it, I had
installed asterisk and confirgured some file, but I can't get a dial
tone on my analog phone.
can someone help?
Regards
Leo
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Why copy...use the make command(same with asterisk)...
make config
Will do all that for you.
DP
On Sun, 2004-05-02 at 22:32, Scott Weis wrote:
Simple solution on redhat machines
In the zaptel source tree (At least the CVS one) there is a file called
zaptel.init. This is a script
All,
I'm having some issues with an X101 clone. The machine is not plugged
into the network right now, but I'll pull the configs and send them on
shortly, but they are similar to the sample configs.
Problem 1: ZAP cannot create channel
For some reason, the cards hang after a call. It's
Whenever we have needed any kind of Carrier Access
(Adit) equipment, we have used Suntel Data. We found
them on eBay and have been buying from them ever
since. They have always been good about fast shipments
and provide good used pricing on equipment.
http://www.sunteldata.com/
Brian
$400 new in stock
- Original Message -
From: Jon Brandon [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, May 02, 2004 8:41 PM
Subject: [Asterisk-Users] Adit 600 FXO card
I am looking at buying an adit 600 off ebay but the ones listed only have
FXS cards in them. Does any one
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