reluctant to realize his promise to me. Deadline has been revised and
revised with different execuses. I just hung up with him at his home
number 1 828 891 5102 and was told that many people/customers showed
up with more money for the code, too much work and the job was
underpriced.
Regards,
Vincent
in in Timbuktu, Ontario while he told me that he lives
in Asheville, North Carolina but home number is a Hendersonville, NC
phone number. I was just updated that according to the phone company
records that is not the name of the person the phone number is
associated with.
Regards,
Vincent
On Wed
explain
the exact reason you do this to me.
People can read and understand what I have been doing here. There will
be NO more from me.
V.
On Fri, 18 Mar 2005 01:49:38 -0500, Brian Capouch [EMAIL PROTECTED] wrote:
Vincent wrote:
Hi all,
You don't want to be fooled by - -. This guy has
Hello
Now that I received an OpenVox PCI card
(www.openvox.com.cn/products_detail.php?genre_id=9id=28), I'm ready
to try and set up a voice server with Asterisk.
We need the following features:
1. When customers call in, they should hear a voice menu asking them
which software they're calling
On Tue, 09 Oct 2007 01:05:41 +0200, Anselm Martin Hoffmeister
[EMAIL PROTECTED] wrote:
Asterisk can do all of that. Something along the lines of
Thanks a lot for the help :-)
___
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Hello
Has someone used the OpenVox A400P01 (ie. a supposedly
Digium-compatible A400P board with a single FXO module
www.openvox.com.cn/products_detail.php?genre_id=9id=28) successfully?
I've put it in an older PC with a Gigabyte GA-7ZX motherboard, then a
more recent PC with an Asrock
Hello
1. I don't have deep knowledge of either Linux or Asterisk, but I seem
to have successfully installed 1.4 with Zaptel (for support for an
OpenVox PCI FXO card) on a stock Ubuntu 7.04 Server Edition:
dmesg ==
[ 25.990943] Zapata Telephony Interface Registered on major 196
On Fri, 12 Oct 2007 09:42:47 -0800, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
Since you are using the OpenVOX FXO card, don't you need another
module? I'm guessing you'd need wctdm INSTEAD of ztdummy.
Thanks. I've seen it mentionned in some articles, but I'm still in the
dark at
On Sat, 13 Oct 2007 15:26:49 +0200, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
I assume that this is a X100P clone handled by wcfxo .
No, it's actually a clone of the TDM400 card. Thanks a lot for the
help :)
___
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Hello
I've never built an IVR before, so I was wondering if someone
could share some code from their extensions.conf that would perform
some of thoses steps:
1. When a call comes in from the TDM FXO port, answer
2. If no CID, play message No CID available. Please type the number
where
On Fri, 19 Oct 2007 14:16:40 -0700, Charles Alvis
[EMAIL PROTECTED] wrote:
http://www.ngnsky.com/product_info.php?cPath=21products_id=50
Thanks. I'll check it out.
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asterisk-users
Hi
SIP is such a pain to use when NAT is involved that I'm willing to buy
an IAX hardphone for someone who works remotely over the Net and needs
to get calls from our Asterisk server, itself behind a NAT.
Are there good, affordable IAX phones you would recommend?
Thank you.
On Sat, 20 Oct 2007 11:37:56 +0100, Alan Lord [EMAIL PROTECTED]
wrote:
Look back a few hours in this mailing list for the message called
IAX2: Incoming calls answered prematurely [RESOLVED].
I have included most of how I setup a simple IVR. It wasn't that hard to
do and I have only been
Hello
I've been googling for this message, but can't find why
Asterisk sends a warning. The configuration files look similar to
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample
It's a TDM card with just one FXO module on it, and I connected an
RJ11 cable to
Hi
The first step I have to go through when users call into our
IVR is to handle the case where users' PBX hides their CID number. In
that case, I need to have them type their phone number (ten digits).
OTOH, those who call without hiding their CID number are sent directly
to the main
On Sun, 21 Oct 2007 10:59:49 -0400, C F [EMAIL PROTECTED] wrote:
I believe that by reloading without restarting asterisk doesnt reload
the signalling part
Thanks for the help. I did read this somewhere, so I typed stop now
in the CLI, followed by safe_asterisk, asterisk -r, and reload:
I still
On Sun, 21 Oct 2007 17:23:03 +0200, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
chan_zap cannot change signalling of a channel on reload. So that
parameter is ignored on reload.
False warning...
OK. So to check that Zaptel is correctly configured, I can just type
zap show channels in the CLI.
On Sun, 21 Oct 2007 13:31:28 -0400, Doug Lytle [EMAIL PROTECTED]
wrote:
You'll want to look at the Privacy Manager:
Great :-) I'll take a look... once I can get the TDM card to pass the
CID number to Asterisk when it's actually sent by the telco.
Thanks for the tip.
Hello
I've been googling for a couple of days now, but still can't
figure out what to put in zapata.conf to get it to report CID.
Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202
as CID FSK Standard:
http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg
On Mon, 22 Oct 2007 10:14:41 -0400, Jared Smith [EMAIL PROTECTED]
wrote:
Instead of ${callerid} here (which probably isn't working for you
anyway), you probably want to use the CALLERID dialplan function to
retrieve the CallerID number, like this:
Thanks for the tip. It'll come in handy... once I
On Mon, 22 Oct 2007 21:19:27 +0200, Vincent
[EMAIL PROTECTED] wrote:
Does Zaptel support those on Digium TDM400 clones like those from
OpenVox?
Pff, finally found what it was: It had nothing to do with zaptel, and
everything to do with extensions.conf:
exten = s,1,NoOp(Got a call
On Mon, 22 Oct 2007 09:06:00 +0200, randulo [EMAIL PROTECTED]
wrote:
The first ten sites that come up, including voip-info.org, usually a
good place to look first, each have full examples. Look also for the
background application wich is used to play the file, get input and
jump to the extension
On Mon, 22 Oct 2007 16:41:19 -0500, Erik Anderson
[EMAIL PROTECTED] wrote:
Version 2 of TFOT was just released a few weeks ago...
Just had to ask :-) Thanks.
___
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asterisk-users
On Mon, 22 Oct 2007 17:57:44 -0400, Jared Smith [EMAIL PROTECTED]
wrote:
Beginning with Asterisk 1.4, we moved all of the CallerID functionality
from channel variables and applications to a single CALLERID dialplan
function. This should have been noted in UPGRADE.txt. I also tried to
warn you
On Mon, 22 Oct 2007 16:09:39 -0700, Ira [EMAIL PROTECTED] wrote:
try adding a wait(1) right in the beginning, worked for me.
Thanks but I had this before, and it makes no difference. Jared
explained above why CID isn't displayed when using 1.4.
___
Hello
When using LookupCIDName, Asterisk 1.4 says that it's
deprecated, and we should use ${DB(cidname/${CALLERID(num)})}
instead, but I don't know how to use it:
;DEPRECATED exten = s,1,LookupCIDName
;ERROR
exten = s,1,${DB(cidname/${CALLERID(num)})}
I guess I should use this as a
On Thu, 25 Oct 2007 18:46:19 +0200, Vincent
[EMAIL PROTECTED] wrote:
I guess I should use this as a parameter to a function, but which one?
Never mind, I found how to use it:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num
Hello
I'm learning more about dialplans and have a couple of questions:
1. Am I right in understanding that the actions that can be performed
in extensions.conf can be of two types only:
- internal commands (Dial, Wait, etc.)
- calls to external scripts throught AGI?
2. I'd rather write scripts
On Mon, 3 Mar 2008 10:14:02 +0800, NOC Ph [EMAIL PROTECTED] wrote:
This questions might annoyed experts. Please bear with me...
The journey of a thousand miles begins with a single step. Lao
Tzu.
Free PDF of Asterisk: The Future of Telephony, Second Edition
On Tue, 26 Feb 2008 01:19:23 +0200, Atis Lezdins [EMAIL PROTECTED]
wrote:
To help you on your way of minimizing modules, here's some basic setup
that generally works
Thanks much for sharing your modules.conf.
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On Sun, 9 Mar 2008 03:11:58 +0100, Grygoriy Dobrovolskyy
[EMAIL PROTECTED] wrote:
Two ways, use n priority or add 'g' iption in dial command.
2008/3/9, Jim Duda [EMAIL PROTECTED]:
How do I get a context to continue to execute commands after the caller
hangs up after a Dial command? I'm using
On Sun, 09 Mar 2008 17:21:47 -0400, Jim Duda [EMAIL PROTECTED] wrote:
exten = s,1,AGI(MisterHouse.agi,Sphinx Connect)
exten = s,2,Dial(CONSOLE/1)
Unless there's a technical reason for this, you should use n, so you
can easily add/remove instructions without having to renumber
everything:
From
Hello
I run AGI scripts from extensions.conf to save data into an SQLite
database file, but this file must also be accessible in read-write
mode by PHP scripts served by Lighttpd.
As far as I can tell, Asterisk runs by default as root:wheel. I don't
know if AGI scripts also run as
Hello
For testing purposes, is it possible to call an extension from the
command-line interface, just so I can trigger calls to AGI scripts
from a test extension?
Thank you.
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On Sun, 23 Mar 2008 19:55:32 -0600, Chris Carey
[EMAIL PROTECTED] wrote:
Correction: I run the web server and asterisk both as the user asterisk
I wish I could, but I have no idea how to safely tell Asterisk to run
as www instead of root, as it does now. I assume I'll have to
chmod/chown a bunch
On Mon, 24 Mar 2008 11:05:32 -0800, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
If the AGIs do run as root:wheel, then there should be no problem,
because they should be able to access the db files?
I agree, but even after uninstalling Lighttpd and installing Apache2,
just to make
On Mon, 24 Mar 2008 12:09:00 -0800, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
Now, that was run under a webserver. right? not under asterisk as an
AGI? I thought we were expecting to see root:wheel :)
Yup, sorry about: I forgot to say that I use a single SQLite database
to share
Hello
I assume it's possible to do this with Asterisk: To train a new
worker who works remotely, I'd like to have him listen in on support
calls so that he gets to learn the kind of problems that come in, and
how they're solved.
When a call comes in and the support person thinks it's
Hello
Has someone written a web page (preferably PHP) that simply shows what
extensions are currently online?
Thank you.
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asterisk-users mailing list
To UNSUBSCRIBE or update
On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger [EMAIL PROTECTED]
wrote:
http://www.micpc.com/eventmonitor/
Thanks guys. I was also thinking of stand-alone apps like Jabber or
something. The call is simply to know if an extension is on- or
offline.
On Thu, 3 Apr 2008 22:30:10 -0500, kazabe [EMAIL PROTECTED] wrote:
I need connect some LAN stations with SJphone to an Asterisk Server
published on Internet. [...] I dont manage the asterisk server.
I just manage my proxy/firewall, and i need to my users can
connect to that server.
SIP works
Hello
I have a couple of questions about running 1.4.17 on FreeBSD 6.3:
1 .On a FreeBSD host, In modules.conf, I naively removed the following
modules that I thought I didn't need, but after stopping/restarting
Asterisk, Zaptel stops reporting calls:
/usr/local/etc/asterisk/modules.conf
noload
On Thu, 10 Apr 2008 16:14:01 +0300, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
chan_zap.so failed to load as it depends on res_smdi.so ?
I have no idea. Is there an up-to-date list somewhere, or some script
that lists dependencies for each module, so that we have some way of
knowing what can be
On Thu, 10 Apr 2008 11:46:48 -0700, Eugen Soare
[EMAIL PROTECTED] wrote:
So this is just a general question, Is Asterisk really good?
Yes, but you should also look at an alternative that used Asterisk as
a reference (www.freeswitch.org), and make an informed decision.
Hello
This PC had been running a Ports-compiled Asterisk 1.4.16.x
succesfully for almost three months, but this morning, although
Asterisk itself seemed fined, the Zaptel interface stopped taking
calls.
Stopping/restarting Zaptel using /usr/local/etc/rc.d/zaptel stop-start
didn't let
On Wed, 18 Jun 2008 12:47:04 -0500, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Please call the reseller from which you bought the card or the manufacturer
for support.
Will do, although it could be a problem in the Zaptel code, which is
not written by the mfg. Thanks.
On Thu, 19 Jun 2008 11:36:27 +0200, Vincent
[EMAIL PROTECTED] wrote:
Will do, although it could be a problem in the Zaptel code, which is
not written by the mfg. Thanks.
I also notice that I can't restart the driver:
# /usr/local/etc/rc.d/zaptel restart
zaptelkldunload: can't unload file
Hello
I'm running Asterisk 1.4.20.1 on a FreeBSD 6.3 host, and unless I'm
mistaken, it seems like /usr/local/etc/rc.d/asterisk script doesn't
make use of /usr/local/sbin/safe_asterisk to restart Asterisk in case
it crashes.
Is this correct, and if yes, why not use it?
Thank you.
On Tue, 1 Jul 2008 04:23:19 -0700 (PDT), Benjamin Jacob
[EMAIL PROTECTED] wrote:
Care to explain the scenario Vincent?
Is it a SIP peer?
what is the DTMF mode set? etc.
Users call into our Asterisk voice server through a Zaptel PCI
interface from regular phones, usually from a PBX (virtually all
Hello
I'm running Asterisk 1.4.20.1 on a FreeBSD that I compiled from the
Ports collection.
It's the second time I'm having an issue with a FXO card and/or the
Zaptel driver. I couldn't figure out what else to do, so I just
rebooted the server, but I'd like to know what happened, and
On Sat, 06 Sep 2008 12:47:58 -0600, Anthony Francis
[EMAIL PROTECTED] wrote:
If Asterisk is running that will happen. Make sure to shutdown asterisk
cleanly before doing that.
Sorry, forgot to say that I couldn't restart or stop/start Asterisk:
[Sep 6 19:06:17] WARNING[23110]: chan_zap.c:4157
Hello
I updated the Ports collection to compile the latest Asterisk, but
after running make config, make just returns without doing
anything:
=
# pkg_version -v | grep asterisk
asterisk-1.4.20.1_1needs updating (port has
1.4.21.2_3)
^C
# cd
On Sat, 13 Sep 2008 00:44:28 +0200, Vincent
[EMAIL PROTECTED] wrote:
I updated the Ports collection to compile the latest Asterisk, but
after running make config, make just returns without doing
anything:
For those having the same problem: make clean ; make config ; make ;
make deinstall
Hello
Apparently, those are just warnings, but I'd like to know what those
messages mean:
[Sep 19 15:32:43] NOTICE[42559] callerid.c: Unknown IE 18
[Sep 19 15:32:43] NOTICE[42559] callerid.c: Unknown IE 21
Thank you.
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Hello
Here's the scenario in my extensions.conf:
1. Check that CID is available
2. If not, go off-hook, and prompt the caller to type their CID number
3. Whether it was sent directly by the telco or input by the caller,
look up the CID number if the DB, and rewrite the CID name on the fly
4. In
On Tue, 23 Sep 2008 12:23:28 +0200 (CEST), Julien Claassen
[EMAIL PROTECTED] wrote:
I wouldn't know a proper way to check for off-hook. But, couldn't you
change
your dialplan?
Thanks for the suggestion, and this is how the script works now, but
since most customers do call with CID enabled,
On Tue, 23 Sep 2008 12:29:22 +0200, Vincent
[EMAIL PROTECTED] wrote:
Isn't there a way to check the status an FXO card is in?
Apparently, it's OK to call Answer() even if the channel is already
open:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Answer
So I guess I can simplify things
Hello
I'm running Asterisk 1.4.21.2 on FreeBSD 6.3.
This part of extensions.conf...
;play a menu, and expect user to type any extension 1-4 or 9
exten = s,n,Wait(1)
exten = s,n,Background(main_menu)
exten = s,n,WaitExten(5)
exten = s,n,Hangup()
exten =
On Fri, 18 May 2007 08:49:49 +0100 (BST), in
gmane.comp.telephony.pbx.asterisk.user Gordon Henderson
[EMAIL PROTECTED] wrote:
Yes. You need to do a few things. Firstly, you need the asterisk server on
a static IP address on the inside, so make sure it doesn't get it's IP
address from the local
On Tue, 15 May 2007 15:52:44 -0400, in
gmane.comp.telephony.pbx.asterisk.user you wrote:
Freepascal seems to work very nicely. However, I'm not sure how delphi
behaves with stdin/stdout since I've not written many console apps in
delphi, mostly GUI rich software. The best bet would be as
On Wed, 16 May 2007 12:17:05 +0300, in
gmane.comp.telephony.pbx.asterisk.user Diego Iastrubni wrote:
This will be VERY slow. Other options might be writing to the asterisk socket
(I heard it's not that reliable). But again, this will be a problem on remote
scenarios.
What I have been using is
I really appreciate your help :-)
On Mon, 21 May 2007 10:15:40 +0200, randulo [EMAIL PROTECTED]
wrote:
What happens when you do the echo test, call it from each phone?
Cool, I didn't know about Echo() .
I added extension 111 from this example:
http://www.asteriskguru.com/tutorials/echo.html
On Tue, 22 May 2007 11:45:34 +0200, randulo [EMAIL PROTECTED]
wrote:
Upon replacement of the Linksys, everything worked fine except audio
on the Sipura. Turns out you need Symmetric RTP turned on in the phone
as Chris Mason says below.
Thanks for the tip. The IP phone doesn't have a setting that
On Wed, 23 May 2007 10:43:04 -0400, in
gmane.comp.telephony.pbx.asterisk.user you wrote:
Has anyone implemented Asterisk behind D-Link Router?
Got one pain in butt customer who wants to setup * system behind D-Link
router model DI-624?
sip.conf:
[general]
externip =
On Tue, 29 May 2007 07:39:40 -0400, in
gmane.comp.telephony.pbx.asterisk.user Luis Morales wrote:
# send the result over callerid ;-)
$AGI-exec('SetCallerId', $response-content);
$AGI-exec('Dial', $ext);
$AGI-hangup();
I'm sorry, but I don't understand why you added this in the script
that
On Tue, 29 May 2007 10:23:18 -0300, in
gmane.comp.telephony.pbx.asterisk.user Gustavo Cordeiro wrote:
No, but I think that you can't install this OpenVox board in this
NetStation case, because the card is a full length PCI and the PC case
supports only half length PCI cards.
Thanks guys for
On Fri, 1 Jun 2007 14:46:14 +0800, in
gmane.comp.telephony.pbx.asterisk.user you wrote:
The Openvox A400P01 is not a full length PCI card. It's a half-length PCI
card. You may be referring to the Openvox A1200P (12 port) and that is a
full length card.
Yup, that's what I figured by looking at the
Hello
Since the Ports collection showed that there were more recent
versions of Asterisk and Zaptel, I tried to compile/install Zaptel,
but it fails, even after stopping Zaptel cleanly, and even after
stopping Asterisk itself, so I decided to just reboot.
Now, when I type ztcfg -vv, I
Hello
I'm contemplating building an Asterisk voice server out of the compact
Asus EeeBox:
http://www.asus.com/products.aspx?l1=24l2=165
But they're so compact, they don't have a PCI slot to handle an analog
phone line. I'd like to minimize footpring and cables: Besides
analog/SIP boxes like
On Tue, 06 Jan 2009 16:51:40 +0100, Loic Didelot
ldide...@mixvoip.com wrote:
Use xorcom products: www.xorcom.com
They provide usb devices for: fox, fxs, bri, pri
Thanks but apparently, they don't have single-line USB devices, just a
whole bank:
Hello
For those SOHO customers (ie. at most, a couple of POTS/ISDN
connections and simultaneous SIP calls) who'd rather not use a big,
noisy PC to run Asterisk, I'd like to offer an alternative that has
the following features:
- not old hardware sold on eBay, ie. it must be up-to-date hardware
Hello,
On Thu, 23 Apr 2009 05:18:27 -0500 (CDT), Joe Greco
jgr...@ns.sol.net wrote:
Can you give us some clues as to why you have disqualified the fanless
and/or embedded devices that are normally recommended on the list
(Soekris, etc)?
I haven't: I'd like to know what the options are. I'm
On Thu, 23 Apr 2009 11:51:02 +0100, Steve Howes st...@geekinter.net
wrote:
http://tinyurl.com/df8qfm
www.voip-info.org/wiki/view/Asterisk+embedded+systems
Thanks Steve. I knew about this list, but I wanted to make sure there
weren't other, more complete sources about the subject.
So at this
On Sun, 26 Apr 2009 12:51:01 +0100, Tim Panton t...@westhawk.co.uk
wrote:
I'm running asterisk 1.4 on an NSLU2 , only a couple of channels
and minimal transcoding, but it seems fine and stable. £80 + usb storage
Thanks guys for the tips on EdgePBX and the Linksys.
Is the NSLU2 still sold,
On Mon, 4 May 2009 10:07:06 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Do you want to build your own?
If so, you can put togther a 1GHz fanless VIA miniITX board, case (that
will take a drive or flash IDE), memory and psu for well under £200. Same
system has one PCI slot
Hello,
I'm looking for a dirt cheap solution for SOHO use to handle at most
a couple of POTS lines, and I notice that X10?P cards go for $15 on
eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.
I have a couple of questions about those cheap FXO cards:
1. Are they
On Wed, 06 May 2009 14:02:20 +0100, Alan Lord (News)
alansli...@gmail.com wrote:
For a cheap backup to your VOIP service they do the job. I wouldn't use
them for a proper system though.
Thanks for the feedback. I have two more questions:
1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel
On Wed, 6 May 2009 12:17:44 +0200, randulo spamsucks2...@gmail.com
wrote:
Those reading the thread amy be interested in Askozia pbx
http://www.askozia.com/pbx/
Thanks for the link.
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On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Some X100P cards (e.g.: those that are based on SI3034, but not those
basedon SI3035) support programmable impedance settings. Sadly the
wcfxo driver does not support it.
Fixing it should mostly be a matter of
On Thu, 07 May 2009 10:16:55 -0400, Jon Pounder j...@inline.net
wrote:
yeah I agree with the above - I never really found echo to ever be a
problem, my only complaint was on some less than stellar cpu's I was
having dtmf recognition problems.
BTW, can someone explain to a libart major like me
On Thu, 7 May 2009 13:40:20 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Another thing: their global-line-standard should basically (if
properly written) resolve http://bugs.digium.com/view.php?id=11057 .
Though I guess the new code will actually be in DAHDI, as Zaptel is
frozen.
Ah yes,
Hello
I'm thinking of selling an Asterisk server based on Atcom's IP02
solid-state unit with one FXO and one FXS ports:
http://atcom.cn/En_products_IP02.htm
By default, this unit based on a 400MHz Blackfin 532 chip only has
64MB RAM and 256MB of NAND flash. Those can be increased to
On Mon, 01 Jun 2009 10:40:56 +0100, Alan Lord (News)
alansli...@gmail.com wrote:
Check out the Astfin project (http://blog.astfin.org/?page_id=2). I'm
guessing they have already done what you need...
Thanks guys. The LAMP is only used to let the user see the call logs,
so I just need PHP + DBMS
On Mon, 1 Jun 2009 13:21:57 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
You may save yourself a lot of hassle just storing the CDRs in a plain
text CSV file (which asterisk does for you), then parsing it with PHP
directly.
Thanks for the tip. I'll see if I can do without an
On Tue, 02 Jun 2009 02:17:02 +0800, Steve Underwood
ste...@coppice.org wrote:
Linux is not a given here. The Blackfin runs uCLinux, as it has non MMU.
Don't get too enthusiastic about putting complex applications like
Apache, MySQL or PHP on one of those boxes. The memory management
limitations
Hello
I noticed a small bug in the way my extensions.conf work:
Users can choose extensions 1-4 or 9 to tell why they're calling, and
I'll send an e-mail to the person(s) to whom is involved. Extension 4
is actually for personal messages for User1, and extension 9 is for
everyone (User1, User2,
i am using X100P on RHEL4, all incoming calls doing
well, during any outbound call from sip to pstn, it
hangup right away when the remote side pick up the
phone.
i've been trying to trace out this problem for 2days.
for the log snapshot below,
DEBUG[2401]: Exception on 15, channel 1
DEBUG[2401]:
Hello
I just read the 2nd edition of Asterisk - The Future of Telephony.
It's a bit light on using * and Jabber. Can you give me examples of
what we can do?
Thanks.
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Hello
Instead of using PrivacyManager, I'd rather use my own
dialplan to prompt the user for a ten-digit number if they called
while blocking CID.
This code does prompt the user, but
1) hangs up if the user didn't type the ten digits before the timeout
2) if the user did type the right
On Fri, 9 Nov 2007 06:56:11 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Actually, it DOES return, but because you have no further instructions
and since autofallthrough is set to yes, it hangs up at that point.
OK, makes sense.
exten = 777,1,Set(CALLERIDNUM=${CALLERID(num)})
exten =
Hello
About Record(), ATFT 2nd Edition says that if the filename
contains %d, these characters will be replaced with a number
incremented by one each time the file is recorded.
Problem is, the documentation doesn't explain how to refer to this
filename later in the dialplan :-/
In this
On Sat, 10 Nov 2007 21:16:44 -0400, Baji Panchumarti
[EMAIL PROTECTED] wrote:
TrySystem is passing the cmd to (bash) shell, just give it a file match
skeleton as long as you don't have other msgNNN.wav files that
shouldn't be moved.
Thanks, but it won't do, as I need to get the exact filename
On Sat, 10 Nov 2007 23:05:47 -0600, Eric \ManxPower\ Wieling
[EMAIL PROTECTED] wrote:
Why not use ${UNIQUEID}?
It's not listed in ATFT, even 2nd ed, so I didn't know about it.
Seems like ${UNIQUEID} is generated with each new call, and includes
an extension:
-- Executing [EMAIL
On Mon, 12 Nov 2007 09:58:50 + (UTC), [EMAIL PROTECTED]
(Tony Mountifield) wrote:
I'm a little surprised at the variety of band-aid suggestions that have
been posted. All you need to do is refer to show application record,
and you uwill see that the generated filename is available by using
On Sun, 11 Nov 2007 11:18:30 -0600, Eric \ManxPower\ Wieling
[EMAIL PROTECTED] wrote:
You need to look at the files in /path/to/src/asterisk/doc (or /docs, I
don't recall) there is much information there, including a file named
README.variables (1.2) or channelvariables.txt (1.4).
Will do.
On Sun, 11 Nov 2007 13:16:35 -0400, Baji Panchumarti
[EMAIL PROTECTED] wrote:
you can generate your own name using a combo of
STRFTIME() CALLERID() CDR() ( and RAND() if you like )
Thanks for the tip. That's what I'll end up doing, as the filename is
more descriptive than just using a
Hello
Since SIP is a bit of a pain to use with NAT firewalls in the
way between clients and *, I'm considering IAX for soft/hardphones.
One thing though: Does the client have to also use UDP4569 as its
source port when connecting to * on UDP4569, or can the client use any
UDP port
On Sun, 18 Nov 2007 10:49:02 -0600, Eric \ManxPower\ Wieling
[EMAIL PROTECTED] wrote:
The source port should not matter.
Good to know. I'll give ZoIPer/Idefisk a shot then. Thanks.
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Hello
Now that I have my first IVR up and running, I'd like to have Asterisk
create tickets in a bug tracker every time a call comes in. It's a
nice way to know who's calling and why, before following up on the
cause for the call.
There are tons of bugtracking apps out there. Do you know of some
Hello
I didn't find the answer in the ATOF 2nd Ed: When using the Record()
application, I need to know how it ended: Did the user leave a
message, or did he hang up?
If the latter, Asterisk stops right there, while I need to run some
other commands before hanging up:
exten =
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