On Mon, 11 Feb 2008 00:24:14 +, Cheikhou DIAW
[EMAIL PROTECTED] wrote:
i've been googling all night looking for a tutorial that shows how to make
an asterisk standalone voicemail server , no way !
Asterisk: The Future of Telephony, Second Edition
Hello
When a call comes in, I'd like to fork a Python script that
broadcasts a message so that users see the CID name + number pop up on
their computer screen, and simultaneously ring their phones.
The following script doesn't work as planned: It waits until the
script ends before moving
On Wed, 13 Feb 2008 10:59:38 -0200, Diego Aguirre
[EMAIL PROTECTED] wrote:
try to use System() instead of AGI()
Thanks, but no go. I get an error:
[Feb 13 21:57:55] WARNING[2138]: app_system.c:107 system_exec_helper:
Unable to execute '/tmp/netcid.py|2000|Joe'
On Wed, 13 Feb 2008 14:25:52 +0100, Michiel van Baak
[EMAIL PROTECTED] wrote:
If you want it to detach the program from it's parent you
need the double fork yes.
Thanks for the confirmation, but when doing this, the NetCID
application no longer pops up, regardless of whether I put the NetCID
code
On Wed, 6 Feb 2008 20:12:21 +0100, randulo [EMAIL PROTECTED]
wrote:
the phone referred to that Jared mentioned is the Allnet
7960. I have an ongoing review of it here (meaning I never finished it
properly).
Thanks for the tip.
___
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On Wed, 6 Feb 2008 20:12:21 +0100, randulo [EMAIL PROTECTED]
wrote:
http://food4wine.ning.com/
BTW, we also want to receive call notifications on our cell phones. In
addition to using SMS, we found a cheaper alternative which is to use
iMode cellphones and subscribe to Bouygues Telecom's
On Thu, 7 Feb 2008 11:03:22 +, Tim Panton [EMAIL PROTECTED]
wrote:
Is that some form of push notification?
Yup, it comes with the same push mail feature found in BlackBerry.
Much cheaper than either sending SMS's or taking a 3G subscription.
Can't wait for Wimax or cellphones over TV
On Tue, 5 Feb 2008 13:56:37 + (GMT), Tim H. Panton
[EMAIL PROTECTED] wrote:
Jared was talking about a decent IAX hardphone on this list a week or so back,
I don't recall the make.
Google didn't return anything with Jared IAX in the
gmane.comp.telephony.pbx.asterisk.user archives.
You should
Hello
I need to hook up someone's remote PC onto our Asterisk server over
the Net. There are firewalls on each side, so I figured it's time to
give IAX a try, and see if it's less of a pain to use than SIP. And
since IAX hardphones are pretty are, I guess I'll go softphone.
Apparently,
On Sun, 27 Jan 2008 09:09:59 -0500, Lee Jenkins [EMAIL PROTECTED]
wrote:
Sorry, I don't have a sample for you as I write mostly in Freepascal/Lazarus
these days and use my own library for AGI/FastAGI. That said, did you try
saving the file to a fully qualified path?
My hero! :-) That did it.
Hello
I've read in the documentation that we should use the CLI
version of PHP when using it to write scripts called by AGI, because
the prepended HTML bits would confuse Asterisk when it got a reply
from the script through STDIN.
And still, the following works OK, although the CLI
On Mon, 28 Jan 2008 12:01:43 +0530, [EMAIL PROTECTED] wrote:
I have installed asterisk.When I start asterisk it starts normally and shows
the status running.
My partner also installed asterisk. I registered 1 user of her server and 1
user of my server in X-lite.
I am able to call or receive
Hello
I'm curious about what can be done when using Jabber with Asterisk.
What are good examples of this combination?
Thanks.
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To UNSUBSCRIBE or
Hello
I'm pretty much a newbie when it comes to C, but I have to use
this language to write a couple of AGI proggies because I need them to
be statically compiled.
Strangely enough, Google didn't return much when looking for the
Hello, world! of AGI in C.
The following doesn't work: The
Hello
Before I bother calling a PHP script through AGI just to read a number
and rewrite the CID name... I was wondering if Asterisk could be
configured so that DB() uses a SQL server instead of the usual
BerkeleyDB?
;rewrite CIDNAME if found in DB
exten =
Hello
There's a lot of information on VoIP at www.voip-info.org ...
but there's also a lot of outdated information there as well :-/
Since SIP is a pain to use when NAT is involved, especially when both
the Asterisk server and the remote phones are behind NAT... I'd like
to try IAX to
On Wed, 16 Jan 2008 18:08:23 + (GMT), Gordon Henderson
[EMAIL PROTECTED] wrote:
However, you'll need to do similar things to your asterisk box router if
it's behind NAT for IAX as you do for SIP. (You will need a static IP
address on the NAT router and port-forward 4569 to the asterisk box,
On Wed, 16 Jan 2008 12:10:35 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
No, it cannot. You could use func_odbc to formulate your own queries,
though.
Thanks. I don't like ODBC, but if it's stable and not a pain to
install/use, that could be the solution.
Otherwise, there's a new solution
On Sun, 06 Jan 2008 10:20:38 +0100, Vincent
[EMAIL PROTECTED] wrote:
cc1: error: unrecognized command line option -Wno-pointer-sign
*** Error code 1
I wonder if maybe the people who ported Asterisk to FreeBSD aren't
using a more recent version of GCC than what's available in the 6.2
ports
On Tue, 08 Jan 2008 13:43:50 -0500, Jared Smith [EMAIL PROTECTED]
wrote:
I always find that looking at the files that are generated
under /proc/zaptel is very enlightening as far as showing what the
zaptel drivers are seeing.
Thanks for the tip.
___
On Tue, 8 Jan 2008 20:29:20 +0200, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
This change is simply due to different versions of Zaptel. Zaptel =
1.4.6 prints to configure because this message is printed (and has
always been prinetd) before the configuration is actually applied.
Good to know :-)
In
On Wed, 9 Jan 2008 12:05:34 +0200, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
wcfxo is not needed.
Basically all you need is:
modprobe your_card_s_driver
This also pulls all of its dependencies (e.g: zaptel)
modprobe wctdm
Thanks, but on AstLinux, the modules are not unloaded:
===
pbx
On Wed, 09 Jan 2008 06:01:32 -0600, Darrick Hartman (lists)
[EMAIL PROTECTED] wrote:
But look in your /etc/rc.conf file for the ZAPMODS variable. You should
have that variable set to:
ZAPMODS=wctdm
Yes indeed:
#ZAPMODS=wctdm
Should I add this module here, or in rc.modules?
Are we positive
Hello
Since TDM cards are known for being particular when it comes
to motherboards (PCI 2.2, etc.), I was wondering if there is a utility
that can check that the Zaptel driver works OK and can tell if the TDM
card is compatible?
That way, if an FXO module is not reporting an incoming
Hi
On an old IBM Netvista thinclient, the TDM card doesn't detect
incoming calls, although the card seems to be detected, and correctly
configured:
pbx asterisk # lspci
00:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
# cat /etc/zaptel.conf
fxsks=1
Hello
I'm trying to build the Zaptel ports on FreeBSD, but it fails:
==
# cd /usr/ports/misc/zaptel/
# make install
=== Building for zaptel-1.4.6_3
make -C zaptel all
Warning: Object directory not changed from original
/usr/ports/misc/zaptel/work/zaptel-bsd-1.4.6/zaptel
cc -O2
On Fri, 04 Jan 2008 16:39:44 -0600, Bob Smither [EMAIL PROTECTED]
wrote:
After some frustrating times, I began to suspect the cards. Doing more
careful testing, keeping track of the cards, confirmed that two of four
cards I was trying had problems.
I had the same issue with three brand new,
On Wed, 2 Jan 2008 14:11:29 +, Tim Panton [EMAIL PROTECTED]
wrote:
There is a Java Applet I wrote that plays GSM files at :
I'll take a look. Thanks.
___
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asterisk-users mailing
On Wed, 2 Jan 2008 16:52:57 +0200, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
What do you have to gain from using a flash player?
By default, unless a plug-in was installed (eg. QuickTime, yuck), if I
click on WAV file in FireFox or IE (Opera is OK), it spawns the
application that registered with
Hello
Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...
www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
... I recorded a
On Tue, 1 Jan 2008 17:23:29 +0530, Godson Gera [EMAIL PROTECTED]
wrote:
s,2,Playback(/usr/local/lib/asterisk/test_wav_out)
And asterisk will automatically pickup the file that it can play with any
asterisk supported format from the specified path.
OK. Is there a way to tell Asterisk which codec
On Tue, 01 Jan 2008 16:10:47 +0100, MatsK [EMAIL PROTECTED] wrote:
The codec is specified (for a sip device) in sip.conf, like this:
Good to know. Actually, I'll have Asterisk save voicemails as WAV and
move the files to the www's htdocs, and send an e-mail to users with
the link they'll just
On Tue, 01 Jan 2008 11:27:54 -0500, dave cantera
[EMAIL PROTECTED] wrote:
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the
current directory...
Thanks for the script. I'll keep it handy.
On Tue, 1 Jan 2008 21:05:11 +0530, Godson Gera [EMAIL PROTECTED]
wrote:
Asterisk automatically takes care of saving CPU issue as it picks the file
that have less translation cost
Yes, but that's OK for files that I use in the IVR, but not for
voicemail messages. The CPU is too slow to handle
On Mon, 24 Dec 2007 14:27:10 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
It seems strange to make this comment (i.e. higher uptime) in a
conversation about porting zaptel to windows.
I don't think it is. I wouldn't use Windows for big iron, but provided
the hardware + drivers are reliable, and
On Sat, 22 Dec 2007 21:01:47 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
I'm not at all certain what you need to change on the hardware, but it
seems to me it should be trivial. Perhaps something in the BIOS?
I was looking at ways to boot it up over the network, and keep
everything in RAM,
On Sun, 23 Dec 2007 06:15:22 +0200, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
Yes. I have a version of our CD that boots from PXE. It took minor
changes and rebuilding as a PXE image, as Debian Live has basic
support of that already. For simplicity I figure you'll be after a
system that has
On Sun, 23 Dec 2007 15:30:50 +0100, Hans Witvliet [EMAIL PROTECTED]
wrote:
Seems what you write is somehow misleading
I meant that I ordered some CF cards, but until they get here, and
since this baby can boot off a remote server with PXE, I was looking
for a PXEd Asterisk that I could use to
On Sun, 23 Dec 2007 10:10:33 -0500, Ulexus [EMAIL PROTECTED]
wrote:
Compare, too, the respective access times between flash and RAM, not to
mention the write session limits of flash (though again, to consider
this is to make another mountain out of a mole hill).
I'll probably put Linux + Asterisk
On Thu, 13 Dec 2007 20:40:08 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
One of the major advantages of using voip is that call termination and
DIDs are wholly separate matters. You can send outbound calls to
various ITSPs based on least cost routing, leaving your POTS lines free
to take
Hello
Since I got the IBM Netvista to boot Linux, and am still waiting for
the Compact Flash cards that I ordered, I was wondering if someone
knew of an Asterisk distribution that can run on that kind of diskless
host?
I've taken a look at AstLinux and AskoziaPBX, but they both seem to be
meant
On Fri, 14 Dec 2007 15:47:38 +1100, Paul Hales
[EMAIL PROTECTED] wrote:
Umm - you could just buy a SPA-3000/3102/3666/etc.
Thanks but I prefer PCI cards. Less cables, less power units that can
burn, less mess :-)
___
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On Fri, 14 Dec 2007 10:51:10 -0500, Lee Jenkins [EMAIL PROTECTED]
wrote:
I have to reboot my desktop xp box daily for it to run well.
I haven't rebooted my XPSP2 in months, and I let it run 24/7, with a
bunch of apps open at all times. And this is a 300E no-name box.
If your PC is so unstable,
On Fri, 14 Dec 2007 10:30:46 -0700, [EMAIL PROTECTED] wrote:
That said, consider the potential market size for people, the DIY sorts,
who would have Asterisk in their homes.
Precisely: The home/SOHO market is huge, and providing an IVR + PCI
card combo for Windows for, say, $200, would probably
and mutiple extensions, then the
extensionss can make/receive PSTN call simultaneously, is this the
same senerio as the one single POTS line to FXO and multiple
extensions on FXSs?
Thanks for help.
Vincent
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On Fri, 14 Dec 2007 17:34:04 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
Yes, the market is potentially huge...for a packaged solution.
If all it takes in plugging the PCi card in their PC, and running
setup.exe, it's no worse than installing a printer. I would imagine
that the standard
Hello
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Is the Zaptel/Zapata combo too *nix-centric?
Thanks.
___
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On Wed, 12 Dec 2007 21:42:49 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
You can keep the POTS line but remote call forward to your ITSP.
Yup, but
1) the telco that handles the POTS line charges us for the connection
between our POTS number and the ITSP, with the caller obviously paying
for
On Thu, 13 Dec 2007 22:21:50 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
It is likely to be a very strenuous job to port the framework and all of the
drivers.
Too bad, because there doesn't seem to be any PCI card for FXO/FXS
available for Windows.
On Fri, 14 Dec 2007 14:50:28 +1000, [EMAIL PROTECTED] wrote:
Erm, there just might be, take a look at this...:
Ah yeah, forgot about $angoma ;-) I'll restate this as: No card for
home/SOHO use, ie. in the $50-100 range for the single FXO port model.
On Tue, 11 Dec 2007 20:34:50 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
Then I migrated to a Soekris Net4801 and dropped that FXOs completely.
I must say that for me that was a good decision. For about 6 months I
call forwarded my numbers to DID provided by an ITSP. I actually tried
several
Hello
I'm looking at my options to build a compact, silent, headless
Asterisk server to handle one or two FXO ports. Out of curiosity, I
got one of those babies on eBay for 20E:
http://silicon-verl.de/home/flo/software/netstation-8364/
Before I spend time on this, can someone tell me
On Tue, 11 Dec 2007 10:09:34 +, Chris Boczko
[EMAIL PROTECTED] wrote:
Im just dipping my feet into the asterisk world, and im having major
fxo problems
I'm no Asterisk expert.
The X100p is a cheap clone i got off ebay for a tenner, so im not
expecting much, i know they have echo issues, but
On Tue, 11 Dec 2007 11:05:24 -0600, Carlos Chavez
[EMAIL PROTECTED] wrote:
The only thing you need to do is set nat=yes when you configure the
phones in Asterisk. You may need to use a STUN server in case the
phones do not properly see the outside address. Once the phones
register they
On Tue, 11 Dec 2007 19:24:52 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
There's no reason why that could not work for you. With a 266 MHz CPU
you have a platform roughly comparable to a Soekris Net4801. That means
limited transcoding.
Thanks for the tip on the HP T5700. There's one for sale
On Sun, 9 Dec 2007 18:15:43 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
No, but you can use func_odbc with a backend SQLite driver.
OK, I'll give ODBC a shot, or call SQLite through a PHP script
instead. Thanks.
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Hello
The DB() application is fine as long as we don't need more than one
value pointed to by a key, ie. the way SleepyCat works.
Problem is, for each phone number, I'd like to map more than one
column, eg. name, e-mail, fax, etc.
Is there a way to have DB() use SQLite instead of AST, or a way
On Fri, 7 Dec 2007 15:12:03 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
You could also think of it as the difference between a procedure and a
function. [...] Unlike other languages, in Asterisk, the return value of a
function
may not be directly ignored (i.e. you HAVE to get it, even if you
On Fri, 07 Dec 2007 15:19:49 -0500, Jared Smith [EMAIL PROTECTED]
wrote:
Hopefully I've explained it in such a way that it's clearer to you know.
If not, let me know and I'll try to be more clear.
Nope, good enough for me :-) Thanks.
___
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Hello
Out of curiosity, what's the difference between a function and an
application?
asterisk*CLI core show functions
asterisk*CLI core show applications
Thanks.
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asterisk-users
Hello
Some of our customers call with CID blocked. I'd like to save
those numbers into a SQLite database using a command-line PHP script,
so that I can...
1. Edit the CID name through a PHP web script which will just list all
the customers in the database who have a phone number but no
On Thu, 06 Dec 2007 05:11:24 +0100, Philipp Kempgen
[EMAIL PROTECTED] wrote:
The line break is not a good idea.
It's not in the script, just my news reader :-)
Not sure about more than one argument. Maybe
Both work. Thanks a lot!
___
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Does someone know why the posts from some users on Usenet are just one
long line, with no carriage return?
On Tue, 4 Dec 2007 17:02:12 +, dadsadsadf dsadasdsa
[EMAIL PROTECTED] wrote:
Hi all, I want to use Asterisk as an IVR system.
O'Reilly's Asterisk, the future of telephony doesn't have a
On Mon, 3 Dec 2007 16:34:39 +0700, Newbie [EMAIL PROTECTED]
wrote:
I am stuck when trying to register SPA-3102 on AsteriskNow ..
I don't use AsteriskNow, but I did use the SPA-3102. The idea is that
you must create an account for it in sip.conf, and configure the 3102
to connect to the * server
On Sun, 02 Dec 2007 23:56:25 +0200, Zoa [EMAIL PROTECTED] wrote:
There are many, (i'm one of the people working for zoiper):
In that case, I think it'd be useful to add a forum on the site, so
people can post when they have problems with the software :-)
Look at the iaxclient homepage,
Thanks
On Mon, 03 Dec 2007 08:14:32 -0900, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
I admit I haven't seen an attractive-enough reason to switch from
straight extensions.conf to AEL for the dialplan.
Thanks. I need to let admins add new items in the database (people who
called with
On Fri, 30 Nov 2007 09:52:59 +0100, randulo [EMAIL PROTECTED]
wrote:
I have used SIP and IAX for about three years now. We don't do a lot
of traffic, but I haven't really seen a difference in quality or
dropped calls.
Sorry for jumping in, but besides ZoIPer/Idefisk, are there
IAX-capable
On Fri, 30 Nov 2007 10:54:47 -0900, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
Sorry it's in some pseudocode that doesn't really represent a language
at all.
BTW, how do most people write dialplans these days? Do they still use
extensions.conf, or did they move to either AEL, AEL2,
On Thu, 29 Nov 2007 23:55:38 -0600, John Faubion
[EMAIL PROTECTED] wrote:
The newer CF cards are making this nearly a mute point. Seems like I provide
updated software often enough that I never have CF cards wear out.
I guess /tmp can live in RAM, but what about eg. recording ten-twenty
WAV files
On Fri, 30 Nov 2007 00:30:06 -0500, Jared Smith [EMAIL PROTECTED]
wrote:
Sounds like a perfect application for the ISNULL dialplan function. Of
course, that adds a whole new set of curly braces and parentheses to
watch out for.
Thanks Jared for the pointer :-)
exten =
On Thu, 29 Nov 2007 00:06:38 -0600, John Faubion
[EMAIL PROTECTED] wrote:
Many of the thin clients fit the bill nicely. I've been using MaxSpeed
MaxTerm clients lately.
Thanks for the tip. It seems like they no longer manufacture them:
http://www.neoware.com/products/hardware/
I'll look in the
On Mon, 26 Nov 2007 21:23:59 -0500, Adam Moffett [EMAIL PROTECTED]
wrote:
This method should work:
${IF($[${STAT(e,/tmp/${CALLTIME}.wav)} = 1]?${CALLTIME}.wav:)}
Yes indeed :-)
===
[internal]
exten = 888,1,Playback(leave_msg)
exten = 888,n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d-%b-%Y-%Hh%M)})
On Sun, 18 Nov 2007 22:14:15 +0100, Giuseppe Barichello
[EMAIL PROTECTED] wrote:
I have successfully compiled and installed Asterisk on an Alix board
(AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian
variant).
Very nice :-)
I'd rather use a PCI card to connect * to the POTS, and
Hello
I've tried a bunch of things, but still get errors/warnings
when using the IF() function:
== TEST #1
exten = h,n,Set(WAV_FILE=${IF($[ ${STAT(e,/tmp/${CALLTIME}.wav)}
]?${CALLTIME}.wav)})
[Nov 26 21:52:34] WARNING[5074]: func_logic.c:107 acf_if: Syntax
On Mon, 26 Nov 2007 23:40:37 +0100, Turbo Fredriksson
[EMAIL PROTECTED] wrote:
What you do is you always write the beginning _and_ the end at once. Never try
to do them
'later'...
Thanks guys. I think I found where it goes wrong:
==
1. /tmp/test.wav exists - the $[] is true:
exten =
On Tue, 27 Nov 2007 00:20:56 +0100, Vincent
[EMAIL PROTECTED] wrote:
Is it a known bug, and does Asterisk 1.4.14 solve this?
FWIW, upgraded from 1.4.13 to 1.4.14, same warning. I guess it's not a
real issue, and I'll just go ahead and ignore it, and find a way to
act on whether the file exists
On Tue, 27 Nov 2007 00:05:41 +, didier [EMAIL PROTECTED]
wrote:
Rpgrade but are you sure a '}' is not missing in your h,n,SET... ?
exten = h,n,Set(CALLTIME=test)
exten =
h,n,Set(WAV_FILE=${IF($[${STAT(e,/tmp/${CALLTIME}.wav)}]?${CALLTIME}.wav:'')})
I don't think so, but it could be.
Hello
I noticed something nasty with the Record() function: If the
user either hangs up during the prompt (ie. doesn't leave a message at
all), or does leave a message but forgets to hit the # key at the
end... Asterisk stops right there, so the rest of the script doesn't
run:
On Fri, 23 Nov 2007 12:38:45 -0500, Baji Panchumarti
[EMAIL PROTECTED] wrote:
Sometimes I use press 1 to leave a msg to reduce the
number of dead air msgs from callers.
Good idea. BTW, making changes to zapata.conf did allow the Zaptel,
and hence */Record to tell if the user hung up during the
On Sun, 25 Nov 2007 19:03:41 -0500, Doug Lytle [EMAIL PROTECTED]
wrote:
What exactly are you trying to do? If a user hangs up during your
Record, it'll go directly the the h extension if it exists.
Ah, didn't know about this extension :-/ I assumed Asterisk would go
on to the next line in the
On Wed, 21 Nov 2007 15:45:35 -0500, Baji Panchumarti
[EMAIL PROTECTED] wrote:
STAT() and record() are doing exactly what they are
supposed to. Use the s flag to fetch the file size. You
have to try a few hangups and figure out a minimum
file size that qualifies as a recording in your setup.
Hello
Some of our customers bought a bunch of phone numbers whose prefix is
the same, eg. 555-12xx - 555-1200, 555-1201, etc. There's a telco
name for this, but I forgot what it's called (think it's DID in ISDN.)
To avoid having to input all those numbers in the DB in the cidname
group, is there
On Wed, 21 Nov 2007 01:29:24 -0800 (PST), bilal ghayyad
[EMAIL PROTECTED] wrote:
Is there a softphone that can be installed on a mobile
(new mobile models), so it can work with Asterisk as
following:
I guess you're really looking for a (smart)phone that supports wifi in
addition to GSM, and to
On Tue, 20 Nov 2007 23:27:34 -0500, Baji Panchumarti
[EMAIL PROTECTED] wrote:
use dialplan function STAT()
Thanks for the tip, but it doesn't seem to work:
==
exten = 888,1,Playback(/root/asterisk_sound_files/leave_msg)
exten = 888,n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d-%b-%Y-%Hh%M)})
On Wed, 21 Nov 2007 12:54:22 -0600, Rob Schall [EMAIL PROTECTED]
wrote:
what cause's this? How do I get just 99?
Maybe there's a better way, ie. making the ISDN card or Polycom unit
handle the presentation, but you could have Asterisk rewrite the CID
name/number on the fly.
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 =
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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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
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 ${UNIQUEID} is generated with each new call, and includes
an extension:
-- Executing [EMAIL
Hello
About Record(), ATFT 2nd Edition says that if the filename
contains %d, these characters will be replaced with a number
incremented by one each time the file is recorded.
Problem is, the documentation doesn't explain how to refer to this
filename later in the dialplan :-/
In this
On Sat, 10 Nov 2007 21:16:44 -0400, Baji Panchumarti
[EMAIL PROTECTED] wrote:
TrySystem is passing the cmd to (bash) shell, just give it a file match
skeleton as long as you don't have other msgNNN.wav files that
shouldn't be moved.
Thanks, but it won't do, as I need to get the exact filename
Hello
I just read the 2nd edition of Asterisk - The Future of Telephony.
It's a bit light on using * and Jabber. Can you give me examples of
what we can do?
Thanks.
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Hello
Instead of using PrivacyManager, I'd rather use my own
dialplan to prompt the user for a ten-digit number if they called
while blocking CID.
This code does prompt the user, but
1) hangs up if the user didn't type the ten digits before the timeout
2) if the user did type the right
On Fri, 9 Nov 2007 06:56:11 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Actually, it DOES return, but because you have no further instructions
and since autofallthrough is set to yes, it hangs up at that point.
OK, makes sense.
exten = 777,1,Set(CALLERIDNUM=${CALLERID(num)})
exten =
Hello
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
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
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