On 16/02/2005 at 09:00 Michael Graves wrote:
Andy Powell has prepared a CF image at www.automated.it/asterisk. I
have been able to get this booted on a testbed system.
Sadly, I'm a Linux newbie and not skilled at command line
administration, thus I'm stuck at the moment. I can get the existing
On 02/01/2005 at 11:21 Michael Graves wrote:
Hi All,
I've read J.R. Richardson's paper Create an Embedded Asterisk Server
which outlines making a Debian server that boots from a compressed disc
image on a CF card. I'm really interested in this as I want my * server
to be more like an appliance
On 09/12/2004 at 09:22 Eric wrote:
Hi Sean,
Thanks for your reply, but that wasn't exactly what I was getting at.
I don't need to increase the system's imposed limit on the number of
open files. I'm more concerned to see if anyone has run across a
memory or fd leak in asterisk that sucks them
Is it April 1st already, where did the year go
Andy
On 25/09/2004 at 01:47 SeshKanuri wrote:
Dear Valued OnDO users,
OnDO PBX v1.3 now supports 100 concurrent calls
Brekeke is
On 25/09/2004 at 14:31 Arik Funke wrote:
Hello together,
I am setting up a communication server which should also act a
very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500
MB usb memory stick. What is the ABSOLUTE minimum space requirements for
~ running asterisk to work as
On 17/09/2004 at 12:21 Pawlowski Julian wrote:
I'd like to create the following: a user picks up the phone
(gets a dial tone), dials '0' for an 'outside' line, gets a
second (different?) dialtone, and is able to enter an
external phone number.
Klaus-Peter Junghanns has something like this on
On 23/09/2004 at 13:36 Joe Antkowiak wrote:
There are quite a number of positive (for end users) implications of
doing this too... just think about all those cell providers that
offer unlimited mobile to mobile calls, and then all those unlimited
LD packages from landline and voip providers.
On 09/09/2004 at 18:48 Josh Roberson wrote:
I wrote cepstral regarding this at the beginning of the week, thought it
might be relevant to post the reply:
Thanks for contacting us. Our Linux package is off the site right now
because we are releasing a new version, 3.02, next week. This is an
On 07/09/2004 at 23:57 Benjamin on Asterisk Mailing Lists wrote:
On Tue, 07 Sep 2004 08:14:57 -0500, Brian Capouch [EMAIL PROTECTED]
wrote:
If you have a Linux laptop with you, then in fact the SIP devices can be
configured to hide behind it. The laptop can then run an instance of
asterisk
On 02/09/2004 at 10:08 Andrew Newton wrote:
Hi,
I have been looking for info on * and the BT Easicom 1000 without much
luck when i found a post to this list from Andy Powell saying that he
had the phone working quite well. Before i go buy a shedload of these
things I would like to know what
On 29/07/2004 at 15:49 Johan wrote:
A very helpful person just sorted the problem out. Apparently, changing
the incoming dial in extensions.conf to Tr solved the problem.
Thanks
...but your caller will get a ringing tone even if your phone number is engaged...
Andy
Brian Elton wrote:
The phone stops working after about 20-30mins if I have
mailbox=context in Asterisk; when I do have mailbox=contect in
asterisk the sip debug returns 481 extension does not exist.
Anyone willing to help me figure out why?
what do you mean :
mailbox=context (or
Hi,
I've just (earlier today) updated from CVS so that I can apply the dtmf caller id
patches. Unfortunately this has had an undesired effect.
I have an intertex ix66 which up until the CVS update allowed me to register my *
server with the ix66 for my local domain (eg sip.mydomain.com). Now
On 17/07/2004 at 20:25 Josh Roberson wrote:
Seth Remington wrote:
I just updated from CVS and noticed that Mark has renamed all of the
parking related files (parking.conf, parking.h, res_parking.c) to
features.conf, features.h, res_features.c respectively. The CVS log
mentions that this is in
On 13/07/2004 at 11:48 Martin List-Petersen wrote:
I can see the point of the discussion somewhere, but let's take it the
other way around (comments though mail):
On Tue, 2004-07-13 at 08:53, Olle E. Johansson wrote:
You have not shown us ANY example yet for which this
facility is *NEEDED*.
For those that don't read every line of source code here's something I found out
today...
Deprecated incominglimit and outgoinglimit
Incominglimit = number of calls the local extension can originate to Asterisk.
Outgoinglimit = number of calls Asterisk will terminate to local
On 11/07/2004 at 18:11 Paul Mahler wrote:
Well, this is certainly getting exciting.
Andy, I took your advice and re-read the RFP. Andy--I don't think you are a
Sorry, I was sleeping when these new emails came in
I've read the other responses which seem to make it pretty clear.. and
I don't think we should let these misunderstandings judge the quality of
Paul's Asterisk book. Even authors need to learn now and then :-)
Can I just point out that the reason I said what I said (see, I can't write)
was because Paul steadfastly refused to believe what we were saying, rather
On 11/07/2004 at 08:42 Paul Mahler wrote:
You are confused about what a SIP session is and what a SIP session does.
SIP, session initiation protocol, controls an RTP, real time protocol,
session between two IP endpionts. The end points have to have unique IP
addresses for the session to run.
On 11/07/2004 at 12:31 Paul Mahler wrote:
The whole point of a SIP registration is to identify a UNIQUE device. You
CAN'T HAVE multiple devices registered as the same SIP device. That's WHY
the last device that registers gets the traffic.
WRONG!
This doesn't have ANYTHING TO DO WITH ASTERISK.
On 11/07/2004 at 20:00 Steven Sokol wrote:
Please forgive me for sending that last message to the wrong list. It was
supposed to go to the Dev list.
Sorry,
Steven
LOL, for me at least - this message arrived before whatever message you
accidentally sent...
:D
Andy
On 08/07/2004 at 22:19 usedcanon wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Harold
Workman
Sent: 08 July 2004 20:15
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
what does that have to do with an
On 09/07/2004 at 13:25 Chris Bond wrote:
On Fri, 9 Jul 2004, Antti Lohikoski wrote:
and No identd (auth) response followed with Closing Link: StiX
(Invalid username [~antti.loh])
Maybe your username is invalid.
Install identd and allow TCP port 113 inbound access and it'll work - if
you
On 08/07/2004 at 08:21 Hall, Eric M. wrote:
I know this is a little off list but I can't think of a better place to
ask this question.
I upgrade the phone to 7.1 and it installed the Universal Application
Loader. Now I'm getting Protocol Application Invalid after it reads tftp
SIP(MAC).cnf
On 08/07/2004 at 18:41 Philipp von Klitzing wrote:
and you'll find a link to the Asterisk Live! CD-ROM.
If you have a moment I guess the list (and certainly me) would be
interested to hear about your experiences with this. :-)
Awww c'mon, it's only 29mb download it and try it for yourself
I'd
Hi Hank,
Working on updating it, and perhaps splitting it into more than one page
Andy
On 04/07/2004 at 17:52 hank smith wrote:
hello andy is your user guide updated?
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 04, 2004 5:24 PM
On 04/07/2004 at 14:53 Steven M. Sawczyn wrote:
Hi, I am very interested in VOIP and telephony in general, although
admittedly, I don't know much about the theories and protocols behind it.
Having also an interest in Linux, I was really excited to come upon
Asterisk. I would really like to
On 16/06/2004 at 22:53 Jay Milk wrote:
You're correct -- I believe I pointed out in my original post that there
is a $200+ difference between a cordless Cisco with/without software.
And that's plain ridiculous. Plus, the phone alone isn't worth $500 in
hardware -- so we're obviously dealing
On 14/06/2004 at 14:53 Jose R. Ortiz Ubarri wrote:
Best mailling list support I've ever read!!! Thanks a lot for your help.
Yes,
unfortunately there are a couple of people on the list who will
a) tell you whatever you are doing is wrong and that they know better
b) but not actually offer any
On 10/06/2004 at 09:04 Dan wrote:
Hi,
- Original Message -
From: Juan J. Sierralta P. [EMAIL PROTECTED]
Cool. It is posible to use the GSM phone as a DIAX headset ? At least
there is posible to transmit audio using Bluetooth.
Unfortunately not, because the GSM phone does not
Hi Dan
On 10/06/2004 at 14:01 Dan wrote:
Hi Andy,
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
Any chance of getting this to work with Nokia phones Dan?
No chance unfortunately..
Nokia does not support the extended AT commands set needed to control phone
keyboard
On 10/06/2004 at 14:40 Angel Diaz wrote:
Hi all,
Is there a way to get the called id (the B number) with AGI perl ?
I know how to get the caller id which is working fine and is just below:
code snip
Thanks in advance,
Angel.
use:
$exten = $input{'extension'};
to get the extension
Matthew,
Dial works on a fall thru principle. Thus:
exten = 555,1,Dial(SIP/1000,30)
exten = 555,2,Dial(SIP/2000,30)
should suit your purpose (not taking into account vm), to add another exten just add
it on the dial 'list':
exten = 555,1,Dial(SIP/1000,30)
exten = 555,2,Dial(SIP/2000,30)
exten
*** BEGIN FORWARDED MESSAGE ***
On 07/06/2004 at 23:34 Andy Powell [EMAIL PROTECTED] wrote:
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Tue, 08 Jun 2004 14:54:33 +0200
Subject: Fwd: Re: [Asterisk-Users] dialplan experts needed
Sorry misread your message, you want
On 08/06/2004 at 11:15 John Fraizer wrote:
exten = 555,1,Dial(SIP/1000,30)
exten = 555,102,Dial(SIP/2000,30)
exten = 555,103,Dial(SIP/3000,30)
exten = 555,104,Voicemail2(u3278)
exten = 555,105,Hangup
exten = 555,2,VoiceMail2(u3278)
exten = 555,3,Hangup
...should be
That's why
On 04/06/2004 at 14:36 James W. Brinkerhoff wrote:
On Thursday 03 June 2004 07:05 pm, Andy Powell wrote:
chan_btp
Hi Brian,
You might also like to take a look at chan_btp and the btp daemon
which allows the use of bluetooth devices to change routing. Since
any old linux box that can handle
This one came up a week or so ago on list... please check the archives before posting.
use 's' before the CALLERIDNUM
ie
exten = 999,2,VoicemailMain(s${CALLERIDNUM})
Andy
On 03/06/2004 at 14:41 Reto Stauss wrote:
When a user dials 999 he is always asked for the mailbox and has to enter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian D'Arcy
Sent: Tuesday, June 01, 2004 2:15 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
Hello all,
I'm going to tackle learning C this week, and
On 28/05/2004 at 19:58 usedcanon wrote:
Hi Andy,
I am most certainly interested. If you have some example code using a DB
(MySQL maybe) that would be extremelly helpful.
BTW, I am new to fpc(Turbo pascal, Delphi and now Kylix), does it have a
linux command line IDE like the DOS version
Thanks
On 01/06/2004 at 11:00 Umar Sear wrote:
Hi Andy,
Once again thanks. This should make things a lot
easier for me. I am greatful.
btw what is the command line to execute the freepascal
ide, also do you have any other recomendations.
Thanks
Umar.
No problem, I hope it comes in handy :D
I
On 30/05/2004 at 22:10 Tilghman Lesher wrote:
On Saturday 29 May 2004 16:53, Andy Powell wrote:
If nobody appears to know, it's probable that they haven't done the
experimentation necessary to show one result or another. If you are
concerned about this behavior, then it falls to you to do
On 30/05/2004 at 21:35 Thor Atle Rustad wrote:
I have just set up my first Asterisk, and I have the basics up an
running.
I have set it up with two SIP phones only. I can call between them, and I
can call out to FWD phones. However, on receiving calls from FWD, my
Asterisk blocks the calls with
On 31/05/2004 at 10:47 Eric Wieling wrote:
On Mon, 2004-05-31 at 10:16, Duane wrote:
Andy Powell wrote:
Anything that's added to * that breaks how protocols work should be by
default OFF not ON,
but that's just IMO...
I agree 100%, this has been very frustrating trying to work out why
*** REPLY SEPARATOR ***
On 31/05/2004 at 11:13 Andres wrote:
Thats the way we prefer it (the old way). Its nice to be able to
publish a sip phone number to anybody out there(for example I can just
say that my number is sip:[EMAIL PROTECTED]). When the
call comes into
On 29/05/2004 at 13:52 brian k. west wrote:
its not really a critical issue... wonder when someone will take the time
and fix it. :P
bkw
to you bkw_ .. it's actually quite important to some of us... a bit like DTMF callerid
:D
Andy
___
On 29/05/2004 at 19:16 Tony Hoyle wrote:
Me too - the current patch could also be used to do DTMF caller ID without
too
much work (there isn't a line reversal in the specs for that, you just
have to
look for valid digits).
I'll probably do some tidying up (change ukcallerid to callerid=uk as
On 29/05/2004 at 16:49 brian k. west wrote:
Accually you can issue the cli commands via manager and get full outputs!
(Most people dont know that)
bkw
yes you can, but you have to have blocking=yes ... and I'm still waiting for info
on what the implications of doing this are.. eg if the
On 27/05/2004 at 22:32 usedcanon wrote:
Hi,
Has anyone done any AGI scripting in pascal. I would appreciate help anyone
can offer. My understandin on AGI scripting is very flaky, I am assuming
whatever language is used the application needs to be compile and made
executable. So if I write a
On 14/05/2004 at 09:00 Olle E. Johansson wrote:
Andy Powell wrote:
I should point out that you don;t actually have to be *using* a ZAP
channel for the background agi to
work. The script starts when the first person enters, once the
conference is over it;s upto the script
to realize
I'm sorry, but any IT Manager who looks upon Internet phoning as a relatively secure
technology doesn't deserve their job.. and any IT Manager that doesn't realise that
VoIP is an IP service and hence susceptible to the pestilence that threatens all
networked systems should be shot where they
On 14/05/2004 at 11:47 Paul Mahler wrote:
Why does voicemail prompt me for an extension instead of just asking my
password?
[voice-mail]
exten = 99,1,VoicemailMain([EMAIL PROTECTED])
exten = 99,2,Hangup
${EXTEN} in your example contains 99 ... you want to use ${CALLERIDNUM}
Andy
Finland, Denmark, Iceland, Sweden, the Netherlands, Belgium, Brazil, Saudi Arabia,
Uruguay,India all use DTMF
So, logically the DTMF solution would be attacked first... but then I do have a bias..
:D
Andy
*** REPLY SEPARATOR ***
On 14/05/2004 at 20:32 Senad Jordanovic
I'd probably shoot him too.. ;)
Andy.
On 14/05/2004 at 13:13 George Pajari wrote:
I'm sorry, but any IT Manager who looks upon Internet phoning as a
relatively secure technology doesn't deserve their job
And what about security specialist Mark Nagil who was quoted
Mitnik is an asshole who used his friends for his own gain...
2600 hertz used to get operator mode
captain crunch whistle generated 2600 hertz tone..
doesn't stop Mitnik being an asshole tho...
Andy
*** REPLY SEPARATOR ***
On 14/05/2004 at 21:02 [EMAIL PROTECTED] wrote:
Have you tried calling Digium sales?
Andy
*** REPLY SEPARATOR ***
On 13/05/2004 at 15:24 [EMAIL PROTECTED] wrote:
Not sure if this is the best place but does any one have any used IAXy's
they are interested in selling?
I am looking to pick one up cheap for a proof of concept
On 13/05/2004 at 14:57 Paul Crick wrote:
I've had a quick look through the mail list and wiki but haven't yet
resorted to looking at the meetme source code.. I see references to a
background agi script that can run if you're using Zap channels. Am I right
in saying that that script runs for each
Ok,
the first think to do is check the permissions on the conf-background.agi ..asterisk
needs to be able to run it ...
The code I've listed below works fine for me:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
%input = $AGI-ReadParse();
$soundpath =
hehehehhe
Yes I know I use cepstral.. I wrote app_cepstral... (bkw messed with it too)
Andy
*** REPLY SEPARATOR ***
On 10/05/2004 at 08:06 Eric Wieling wrote:
On Mon, 2004-05-10 at 05:37, Andy Powell wrote:
I'd love to hear how you get on Ben, but I get the feeling
I'd love to hear how you get on Ben, but I get the feeling that Rhetorical's software
prices are out of the reach of most people here. I think integration of this would be
a very good move tho. Quite frankly Rhetoricals tts is the best I've heard so far.
Andy
*** REPLY SEPARATOR
Ok,
I just thought I'd publicly pat Adrian Kennard (revk) on the back for this
application. This is an excellent contribution and gets my vote for app of the year.
For those that aren't aware app_sms allows you to send/receive fixed line sms messages
from asterisk. ( you can take a look at a
In /etc/asterisk/zapata.conf before the
channel=x
(where x is the channel assigned to the FXO port)
put:
callerid=PSTN Call 1234567
You will need to restart * for this change to take effect
Andy
*** REPLY SEPARATOR ***
On 09/04/2004 at 10:56 Victor Perez wrote:
Is
Just curious, but why does it strike you as such a bad idea? Especially if
it was disabled by default. I can understand you not wanting your system
security or your personal privacy compromised, but I think it would be
great to have it in place for:
A) Manual activation for those who want
On 08/04/2004 at 10:00 John Todd wrote:
Any Day Now(tm). Wasim has fallen off the face of the Earth, but
I've seen with my own two eyes a working copy of the Iaxy from
Digium, so this holds promise. My request for a 1u 24-port IAX-based
box that takes Digium daughterboards (FXO or FXS)
This is a known issue with SIP - look at bug 207 in the bug tracker
Andy
*** REPLY SEPARATOR ***
On 08/04/2004 at 12:37 Scott Laird wrote:
I've noticed a little problem with my setup. I've been using a flaky
version of X-Lite for testing, and it tends to crash every few
I'd like to give this one 10 thumbs down. IMHO a bad idea, a nasty little bad idea..
evil, spawn of Satan. If this were implemented the first job of a new update would be
to rip it out and flush it down the nearest toilet.
I can only wait until we see M$ like activation implemented... oh the
I'd take a look at the VoiceTronic cards ( http://www.voicetronix.com/hda.htm ) which
can be used with * or their free software.. these cards can be configured as :
12 Loop-Start ports only.
8 Loop-Start AND 4 Station ports.
4 Loop-Start AND 8 Station ports (default configuration).
12 Station
This is a fairly simple thing to do. You don;t say what type of phones you are using,
so I;ll assume SIP for the example:
Step 1:
Put
callerid=Darren 1234
for each phone definition in sip.conf, obviously replacing Darren with the user eg
Darren Nay or Joe Bloggs, then replace the 1234 with
Alternatively, put it somewhere where we can all get at it :D
Andy
*** REPLY SEPARATOR ***
On 02/04/2004 at 06:52 Raymond McKay wrote:
I am using one version of their chan_sccp with a 7960, and can vouch for
its functionality there. If you strike out finding an
On 02/04/2004 at 11:17 John Chambers wrote:
Andy Powell wrote:
1 Access to the PSTN - this can be done via a single X100P card (plugs
into a standard phone line) or one of the sinlge port T1 cards or 4 port
TDM410 cards (if you need a shedload of lines). You can also use a VoIP -
PSTN gateway
Please don't tell me you deliberately used LARGE BREASTS as part of the subject for
this...
Adny
*** REPLY SEPARATOR ***
On 31/03/2004 at 18:16 Zot O'Connor wrote:
How do I do this
1) ZAP- * - IAX(1) -- IAX(2) - DG104S -- Handset
2) No Answer on Handset
3)
at 04:09, Andy Powell wrote:
Please don't tell me you deliberately used LARGE BREASTS as part of
the subject for this...
I got got tired of asking questions that did not get answers while
watching people berate dead subjects or each other. The questions have
been thought out, I guess
John,
Yes, asterisk can do that, and in fact it's very simple. The problem at the moment is
your level of knowledge of asterisk, but this can be resolved...
There are a number of things you need:
1 Access to the PSTN - this can be done via a single X100P card (plugs into a standard
phone
] On Behalf Of Andy Powell
Sent: Monday, March 29, 2004 7:16 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call routing based upon callerID
John,
This is referenced as the anti ex-girlfriend feature...
example:
exten = s/12345678,1,congestion
exten = s/24681012,1,Dial(SIP/phone2)
exten = s,1
- Let the caller know its position in the queue (ie: you are number #
in the queue, please hold and an operator will hang on you)
This is not possible at the moment.. Anyone know better?
Actually it is possible have a look at the bug tracker - I would give you the url
but I can't get to
Senad,
I can do better than that:
http://bugs.digium.com/bug_view_page.php?bug_id=214
which says that the patches have been merged into cvs :D
HTH
Andy
*** REPLY SEPARATOR ***
On 30/03/2004 at 17:00 Senad Jordanovic wrote:
Andy Powell wrote:
- Let the caller know its
John,
This is referenced as the anti ex-girlfriend feature...
example:
exten = s/12345678,1,congestion
exten = s/24681012,1,Dial(SIP/phone2)
exten = s,1,Dial(SIP/phone1,30)
also check page 31 of the handbook...
hth
Andy
*** REPLY SEPARATOR ***
On 29/03/2004 at 20:34
Ok,
so I've re-reported a feature request
http://bugs.digium.com/bug_view_page.php?bug_id=0001265
because
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9
was closed for no apparent reason. Is it now policy to simply close off feature
requests when they haven't been added? If it
You could take a look at
http://andreasotto.net/asterisk/
and modify that to suit
Andy
*** REPLY SEPARATOR ***
On 15/03/2004 at 16:46 Tony Wasson wrote:
Darren Nay wrote:
Hello All,
I am just looking into Asterisk as a viable voicemail solution for our
phone
Take a look at
http://www.ainslie.org.uk/callerid/cli_faq.htm
Lots of info there
Andy
*** REPLY SEPARATOR ***
On 14/03/2004 at 11:45 randulo wrote:
Can anyone ell me if they've had experience on the continent with caller
ID on analog POTS lines?
Here in France, we
which it will dial...
Andy
*** REPLY SEPARATOR ***
On 23/02/2004 at 17:26 Peer Oliver schmidt wrote:
Andy Powell wrote:
Snom TAPI integration is a joke...
Would you mind elaborating a bit on this? Is the future implemented, but
does not work, or is it not implemented
Snom TAPI integration is a joke...
Andy
*** REPLY SEPARATOR ***
On 22/02/2004 at 21:47 Peer Oliver schmidt wrote:
Hi,
anyone here running SNOM phones with TAPI integration with Outlook?
Any other hardware phone with some TAPI integration?
rgds
pos
Hi,
can anyone give me any pointers as to how I should configure a snom 100 (with h.323
firmware) to use h.323 between it and *. How can I check that my h.323 install is ok
too.. If i do:
ASTERISK*CLI h.323 show tokens
ASTERISK*CLI h.323 show codecs
I get no info or anything back, if I turn
lo,
Is there a single central location for code and applications other than CVS? I'm
talking about code that can't/wont be included in CVS for various reasons? Does the
wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has
done some updates too) but apparently
Isn't this what the asterisk-addons directory was created for? This
is where the MySQL code was relegated after it became legally
unfavorable to put it in the CVS main branches.
JT
The code in question was actively denied entry into CVS (asterisk core or addons)
Andy
Hi,
I downloaded this the other day and finally got it to stop crashing. It appears that
any response from asterisk
that implies an error (for example dialing a non-existant number, using the wrong
password, selecting a codec
that you've configured a local * not to use etc) resulted in a crash.
*** REPLY SEPARATOR ***
On 27/01/2004 at 15:55 Chris Albertson wrote:
My Asterisk server registers two FWD numbers.
On average I get about one call a day from someone calling
from an FWD number and leaving a pointless, under 10 second
message. It's easy to see who these
If you don't have a fax connected to * then create and exten:
exten = fax,1,Goto(day,s,1)
I had the same today... :/
Andy
*** REPLY SEPARATOR ***
On 15/01/2004 at 16:41 Iain Stevenson wrote:
A caller to me was this afternoon detected as a fax machine:
Jan 15 15:31:17
ok, how about
inside
outside
up
down
server
status
current
is
and finally:
Please look at bug 207
:D
Andy
*** REPLY SEPARATOR ***
On 11/01/2004 at 19:36 John Todd wrote:
Here's the latest batch of words to get shipped out to Allison Smith.
Please submit reasonably small
I'd be nice to be able to play a tone (or message) at AbsoluteTimeout - N where N is a
number os seconds before the cut-off... a bit like pay phones (used?) to do...
eg.
beep beep beep beep beep
click...
Call terminated because you took to long explaining your probelm to the support team,
Nicolas,
I'd appreciate a copy of this if possible... got a url where I can grab it?
Thanks
Andy
*** REPLY SEPARATOR ***
On 09/01/2004 at 10:43 Nicolas Gudino wrote:
Andy Powell wrote:
I'd be nice to be able to play a tone (or message) at AbsoluteTimeout -
N
where N
You can use
immediate=yes ;like the bat phone (old Batman)
in your zapata.conf for the channel... but that means it just gets answered.. I think
your problem is having the operator signal that they can take the call... otherwise
when they get bag from the toilet/coffee break/ciggie break
Hi,
Having installed postgresql-devel-7.4-0.3 and postgresql-libs-7.4-0.3 I'm having
probs. compiling cdr_pgsql, can anyone offer any pointers as to what I might be
missing?
I'm hoping I've just missed out something like postgresql-wibblewobble-7.4-0.3 or
something ...
Below is the result
hi,
since development of dtmf caller id under * is prolly going to only be done if someone
stumps up the cash I've been looking for alternatives... Hoving found a number of
projects which turn out to be mad prototypes or unavailable details i nearly gave up..
then I found this:
Hans although your somewhat right I don't think its fare to ask all
tourists to leave their clothes at customs and to don clogs and ride a
battered old bike around the city. I also must say that from my experience
its very rarely (I've never heard of it) the native Dutch that perform
these
On 17/11/2003 at 18:39 Steve Totaro wrote:
looks like critchy is especially bitchy
With all his whinging, if i didn't know any better, I'd suspect he was using a 2400
baud modem...
Now I'm off to reply a message and change the subject line
Andy
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 (code field not indexed) = 47.183s
Query1 (code field indexed) = 45.731s
Query2 (code field
Your clock is wonky
sync with an ntp server or set the time on your machine...
Andy
*** REPLY SEPARATOR ***
On 21/10/2003 at 15:03 Chris Albertson wrote:
--- Ron Fallara [EMAIL PROTECTED] wrote:
NOTICE[1192484144]: File sched.c, Line 209 (sched_settime): Request
to
From what I see this *IS* a problem with the CVS code...
as a quick fix I suggest using the zaptel code from august 18th 2003 since that is
known to work (I'm using it after having the same problems as you)
It's kinda strange if this isn;t regarded as a bug, as Digium have then EOL'd some of
Yep, it probably will not work with your motherboard. You might try
setting -DNO_CALIBRATION in the Makefile, then running 'make clean
all install' and trying again (this has worked for some people).
Failing that, try it with a different motherboard.
-Tilghman
This is a CODE issue not a
It's kinda strange if this isn;t regarded as a bug, as Digium have
then EOL'd some of their cards and not told anyone, while continuing
to sell them...
Compare revision E to revision C of the card. Revision C is no longer
being sold by Digium.
This may be true, however, they were being sold
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