Ronald Wiplinger wrote:
He want to use 100 phones at the same time!!!
Alas, he won't be able to.
Re: ASTCC in-use flag
You'll have to disable the in-use flag for his account.
--
JP Carballo
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the credits of 100 cards?
In short, an account, not a card. Get my drift?
Or, try making another brand with a markup of 100% I guess. Never tried
that one though.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm
JP Carballo wrote:
Ronald Wiplinger wrote:
I got a request for one customers to set-up 100 accounts.
I use usually the Caller-ID as the card number.
Is there a way to make it for 100 accounts easier?
To generate 100 cards is not a problem, but if it would work with one
account number
to be a
solution?
Nicolas already said and I quote:
Replying to myself... I was thinking on a2billing, not astcc, so
php-pcntl will make no difference.
ASTCC uses Perl-AGI. That should have clued you in.
--
JP Carballo
http://www.netfone2x.com
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It might look like
that to the card's uniqueid.
Hth.
--
JP Carballo
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Maxim Vexler wrote:
I would be interesed in your the applications you use in your dialplan
to reset this flag.
Could you please post here the relevent part of your extensions.conf ?
http://lists.digium.com/pipermail/asterisk-users/2006-April/146947.html
--
JP Carballo
http
your head over this one?
Get the card's uniqueid and use it to check if that particular channel
is up.
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JP Carballo
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Bringing the world closer.
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and the lastest asterisk-perl distrib.
I'm not using 1.2.7.x but I doubt this would change from earlier versions.
P.S. Shouldn't the subject be ANSWEREDTIME? :)
--
JP Carballo
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Bringing the world closer.
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Lacy Moore - Aspendora wrote:
Had to turn my monitor upside down to read them :-)
--
Lacy Moore
Aspendora, Inc.
You must have one of those rotating monitors huh? ;)
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing
with
Virtuemart/Joomla.
There are other applications. Look around.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
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disconnected
-- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected
Any idea?
Someone or something is regularly accessing the manager interface.
Nagios perhaps?
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing
time.
--
JP Carballo
http://www.netfone2x.com
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It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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they are generated
upon card creation, whether or not you use them.
2)you need a PIN because most of the guys
use CALLERIDNUM as cardnumber, and that's open to see. I think it was
JP Carballo suggested earlier to authenticate against another variable
i.e.
It's nothing new as everyone here knows
that only one
card is in use.
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JP Carballo
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argument to astcc.agi
After you've done that, you modify the check for the pin to take the new
argument into account.
You might also want to add a routine to retrieve a substring from the pin.
--
JP Carballo
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Jay Milk wrote:
Could you kindly let us know what numbers those survey-calls will be
coming from, so we can all add them to our blacklists? Thanks!
*snickers*
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular
systems that work with asterisk.
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JP Carballo
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into the
background...
Or like astcc does, use the L argument to the dial command.
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JP Carballo
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.
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JP Carballo
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a similar astcc.agi call for in-network calls.
The cost will depend on how you've implemented TARIFF and whether it
adds to or bypasses the cost field for your 600x SIP route.
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JP Carballo
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It might look like I'm doing nothing, but at the cellular
pressing * to exit and get dropped back into the IVR to
make another call?
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JP Carballo
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?
There could be a call to checkinuse() before the RESET_INUSE routine.
If the RESET_INUSE flag is set, the routine should exit and not proceed
to the following calls to
checkexpired()
checkinuse() and
setinuse()
I suggest you check that the callerid you are using matches a card in
the astcc db.
--
JP
channel =1-32
--
Regards,
Mark Quitoriano, CCNA
What seems to be the problem?
--
JP Carballo
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);
exit(0);
}
And this in extensions.conf:
exten = s,n,DeadAGI(astcc.agi,${CARDNO},RESET_INUSE,2)
I leave it to you to capture ${CARDNO} :)
I don't enable this in the IVR unless the person has entered a valid
account number, for obvious reasons.
--
JP Carballo
http://www.netfone2x.com
the inuse handling mostly intact.
I want it to be anal.
If a customer complains about it, I'm more worried about a trunk failing
than a cheating caller.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really
the whole routine either just before or after these lines.
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JP Carballo
http://www.netfone2x.com
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Jeremy wrote:
Is there anyway for me to authinicate first before givng options
You can add an option to astcc to authenticate the account and then drop
you back out into the IVR.
Or you could use mysql queries in the dial plan to check the account
number before passing it to astcc.
--
JP
.
You could however, store the current time when authenticating.
When the caller hangs up, run another agi that takes this value and
subtracts it from the total time stored by *.
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JP Carballo
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to exit, and is dropped back into the IVR.
If he presses 2, his balance is read back immediately since his account
number is already in a variable.
If he presses 1 again, astcc asks for another number to call.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look
that to astcc.agi
When the person is done with a call, they can press, say, *, exit out to
the menu, dial 1 and be prompted for a new number to call.
The other way is to modify astcc.agi to save the account code to a var
CARDNO.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might
their personal property
and hence the act of not using PLDT's gateways constituted theft.
It was thrown out by the Supreme court of the Philippines.
http://www.manilastandardtoday.com/?page=news06_mar06_2006
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like
to a context with your IVR.
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
himself out of
a loop in emacs-lisp ;)
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JP Carballo
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/mohmp3/*.mp3
Filename: /var/lib/asterisk/mohmp3/fpm-calm-river.mp3
No ID3 Tag
Filename: /var/lib/asterisk/mohmp3/fpm-sunshine.mp3
No ID3 Tag
Filename: /var/lib/asterisk/mohmp3/fpm-world-mix.mp3
No ID3 Tag
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look
) dialplan coding?
Emacs...
On Sat, 2006-03-04 at 01:35 +0100, adibar wrote:
Vim forever ;-)
On Fri, Mar 03, 2006 at 03:06:02PM -0500, S McGowan wrote:
snip
emacs for me :)
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular
seems to wait
for ONE digit! Is there a way to put the calling mobile phone into a context
and wait for a full-length extension?
The documentation on voip-info has just the example you need.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+WaitExten
--
JP Carballo
http
[EMAIL PROTECTED] wrote:
On Monday 06 February 2006 09:25, JP Carballo wrote:
snip
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you
and correct me.
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JP Carballo
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Alex Barnes wrote:
I think the once it's working, leave it alone advice is very sound
indeed :)
A similar rule says If it ain't broke, don't fix it.
--
JP Carballo
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Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm
Jens Vagelpohl wrote:
On 8 Feb 2006, at 09:43, JP Carballo wrote:
Alex Barnes wrote:
I think the once it's working, leave it alone advice is very sound
indeed :)
A similar rule says If it ain't broke, don't fix it.
Until you realize some script kiddie has exploited another Apache
though and all are set to 0.
--
JP Carballo
http://www.netfone2x.com
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Garth van Sittert wrote:
There is a good utility called iaxping to test IAX latency.
Kind Regards
Garth
Another one is check_asterisk.pl plugin from the Nagios monitoring system.
--
JP Carballo
http://www.netfone2x.com
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It might look like I'm doing nothing
ICMP traffic?
You might want to try the tools here:
http://www.dnsstuff.com/
and here
http://centralops.net/co/
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
ram wrote:
Hi
how about SIP friend to SIP Friend
even it taking gsm
ram
Check the [general] section of your sip.conf
Most likely there is an allow=gsm line there.
Just allow=ulaw on your end so you can connect to voipjet.
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JP Carballo
http://www.netfone2x.com
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:
http://www.voip-info.org/wiki-Asterisk+Bootable+CDROM
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JP Carballo
http://www.netfone2x.com
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.
Thanks,.
satish
It is generally trickier to set up dual servers when both of them are
behind firewalls.
See here:
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular
to make call out, its using only GSM
even though i mention g729 in top list
whats wrong ??
ram
IMHO voipjet only allows g.711 ulaw.
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JP Carballo
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Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
not use Authenticate()?
You can then set each person to be under a certain context based on
password or extension.
Under each person's context, include other contexts that allow/limit
their capability to call.
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm
Dean Collins wrote:
Yep I use iax and sip with time warner cable new york.
Works fine.
Dean
IAX2 and SIP used here.
All systems green.
--
JP Carballo
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Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite
see there a query on creating pricelist table, can anyone help me on
this please? Thank You
Regards,
Ronald
Under Rates click on - Pricelists then Add...
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm
now.
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JP Carballo
http://www.netfone2x.com
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though.
I would have one record from the remote server and one from
the local. Correlating one record with another could be a pain.
There are several fields you can use to sync records. You could also set
the accountcode and/or userfield to a value you generate per call..
--
JP Carballo
http
I played with had mp3 capabilities and games and a tiny
thumbpad.
Their extinction was caused by the widespread adoption of prepaid SIM
cards for cellphones and SMS.
It's funny though that the last time I saw a bellbottom there was about
the same time.
--
JP Carballo
http://www.netfone2x.com
for a fraction of the
speed Verizon provides.
Until Verizon is done with their tedious wiring of affluent
neighborhoods, I can only dream.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
Chris Mason (Lists) wrote:
JP Carballo wrote:
I would if I could. I'm paying more to Time Warner for a fraction of
the speed Verizon provides.
Until Verizon is done with their tedious wiring of affluent
neighborhoods, I can only dream.
Rich bastards. Come the revolution...
lol!
It's
.
Customers register through Virtuemart/Joomla, then a card is created
on ASTPP.
When they buy a refill card through the store, their account is credited.
As for * on ser, you may want to visit :
http://www.voip-info.org/wiki-SIP+Express+Router
--
JP Carballo
http://www.netfone2x.com
Bringing
Darren Wiebe wrote:
JP Carballo wrote:
Ronald Ramos wrote:
Hi All,
Any solution on how I can implement prepaid billing on asterisk?
But not the calling card type, just a simple Custome rwill buy
credit, consume then buy again.
Also, is there a solution for that when you combine asterisk
if an order is
confirmed, then the relevant variables are passed on to ASTCC/PP on the
* server.
For further details, feel free to email me.
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
with it, and then post it. Upon hangup (run a
deadagi) and the channel still has the variables in it, so i finish up. It
has worked rock solid.
./ben
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really
://www.yoda.com.tw/model.php?type=Enterprise_VoIPpname=VG400
Anyone had success with this?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
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JP Carballo
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Bringing the world
on the corner of Walnut and Yale in Irvine on the 3rd
Thursday every
month. The following dates are already secured:
Thurs Jan 19
Thurs Feb 16
Thurs Mar 17
Irvine Heritage Park Library
(949) 936-4040
14361 Yale Ave
Irvine, CA 92604
Google Directions: http://tinyurl.com/9vq3e
--
JP Carballo
http
+-+dual+servers
You can use the switch statement on server 203.101.42.14 to make the
server 10.42.16.1's dial plan available.
You can then dial extensions registered on 203.101.42.14 from 10.42.16.1
You cannot use the switch statement on both servers though. Only one.
--
JP Carballo
http
JP Carballo wrote:
Mantu Jha wrote:
Hi I am have two Asterisk server at two different location one is
having static ip 203.101.42.14 and other is having static ip
10.42.16.1 how can i integrate both so that i can use the others dial
plan.
It's all here on this page.
http://www.voip
Tom Lynn wrote:
Their trunk works fine as of the time this email is sent.
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JP Carballo
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roblems. I'm just happy the units I have work well. Thanks to Yoda's
support. They're set on GnuGK rather than Asterisk so it was a first
for me as well as them.
Xie Xie
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing
doing wrong ?
Nothing.
Both of them have never been fully implemented.
--
JP Carballo
http://www.netfone2x.com
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if you Answer() the call before you Dial(), it will switch to the
fax extension.
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
less than the price of their yearly MS license, we would be
glad to hear about it here.
Good luck!
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
started on using Asterisk
in the more advanced modes (ie Realtime) so I do not know how to implement
this, but I'm sure that it could be done.
Ryan
http://dev.mysql.com/doc/refman/4.1/en/replication.html
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JP Carballo
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Bringing the world closer.
It might look like I'm
this:
usblp120960
usbcore111612 4usblp,ehci_hcd,uhci_hcd
Any ideas what I am doing wrong ?
I'll presume you're using 1.2.0
There are 1.2.0 zaptel srpm packages on cooker that build fine.
Haven't tried loading ztdummy itself though.
--
JP Carballo
http://www.netfone2x.com
-info.org/wiki/index.php?page=Asterisk+cmd+System
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JP Carballo
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JP Carballo wrote:
amaury BOSSE wrote:
Thanks for your answer but I don't want to include a file, I only
want to include a variable.
Is it possible to execute linux commands like grep or top in a .conf
file in order to parse a file and get a variable?
Look into the System() command
that you've installed libpq.
That's the package that includes the shared library libpq.so.4, at least
on my system.
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JP Carballo
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Bringing the world closer.
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Bill Michaelson wrote:
Just curious...
Is there anyone out there who has given this outfit money and actually
received any service from them?
I only have a 1800 number from them, but no problems so far.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look
switch in
extensions.conf
It's all there on that page.
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JP Carballo
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multitail) to watch
/var/log/mysqld/mysqld.log
You should see the inserts into the cdr table or lack thereof.
--
JP Carballo
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Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
, (and a few minutes testing), I believe there
was enough info to figure out what was needed.
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
:on6:off
just as they are defined in:
([EMAIL PROTECTED]:asterisk)# grep chkconfig /etc/init.d/{asterisk,mysqld}
/etc/init.d/asterisk:# chkconfig: 2345 40 60
/etc/init.d/mysqld:# chkconfig: 2345 11 90
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm
Eric ManxPower Wieling wrote:
JP Carballo wrote:
Nitesh Divecha wrote:
Hello All,
My Asterisk is configured like a Soft Switch, I have many incoming
and outgoing traffic to different providers.
How can I forward CallerID to my providers. My providers are
complaining big time
--level 2 asterisk off
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
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JP Carballo wrote:
Eric Bishop wrote:
I have:
[EMAIL PROTECTED] ~]# chkconfig --list | grep mysql
mysqld 0:off 1:off 2:off 3:on4:off 5:off 6:off
[EMAIL PROTECTED] ~]# chkconfig --list | grep asterisk
asterisk0:off 1:off 2:on3:on4:on5:on6:off
JP Carballo wrote:
Ok, so? That's an 'If alright. I'm not an oracle. He didn't give us
any examples of what he wanted. I just grabbed the first thing I saw
in my extensions.conf and pasted it.
Granted, that wasn't very bright, but the answer he needs might not
even be what we think. He did
,
Neal
In your provider's outgoing context, add the following before calling out:
exten = _X.,1,Set(CALLERID(number)=${CALLERIDNUM})
exten = _X.,n,Set(CALLERID(name)=${CALLERNAME})
--
JP Carballo
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It might look like I'm doing nothing
codes are you calling? I'll see if I can test them from my end.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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. Is the Asterisk the Gaul comics still in circulation? It's been
years since I read the series...
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
Programmers confuse Christmas and Halloween because DEC 25 = OCT 31.
___
--Bandwidth
schemes are used identify remote machines...
Not a show stopper for me. I have a few screen sessions open for ages.
Sometimes, the only time I see a prompt is when I do M-x shell :)
I'll check out your changes the next time I need to edit a .conf file.
--
JP Carballo
http://www.netfone2x.com
haven't ventured there yet, myself.
Anyway, use and enjoy, comment, or criticize as you feel is warranted.
Big kudos to this list and the * community as a whole. I have learned
a ton in the last two weeks, thanks to all of you.
lyd
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JP Carballo
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Bringing the world
://control.sixtel.net and click on DIDs
You should then see 3 sections:
Numbers assigned to you:
Telephone Numbers (DID)
and
Toll Free numbers
Click on the link under Telephone Numbers to get a DID assigned to you.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look
.
snip
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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betas. I agree that the number
should be set aside then. I wonder what the problem is.
Darren
JP Carballo wrote:
As soon as the number is entered.
Consider the scenario where two people dial in with the same card
number. Once one person has entered a valid number, you want to let
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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]
exten = s,1,Answer
exten = s,2,Dial(SIP/xlite1,10,t)
exten = s,3,Hangup
of course, this is an unconditional forward...
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy
.
Didn't even have to ask then and it has been functioning well since.
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http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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switch.
Is that undocumented?
The wiki entry for the Dial() command says:
w: Allow the /called/ user to start recording after pressing *1 or what
defined in features.conf (Asterisk v1.0.x)
To add a short delay to one of my remote FXO lines, I added A(silence/2)
to the dial string.
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JP
? Is there possibly
something wrong with my configuration
Steven Hall .
[EMAIL PROTECTED]
What does the CLI say?
Does it show Playing 'value-of-MES' (language 'en')?
I'm using 1.0.8 here and I have no problems using A(x) in my dial
strings in either ZAP or SIP channels.
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JP Carballo
http
have replied to the wrong article.
That's his signature.
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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: None
stunserver : None
stun_call is off
alt_registrar : None
allow_num :
None.
Console
Call in from 202/203/204, dial 0112011234567 to route the call to *,
(optionally do whatever you want) then have * pass 1234567 to 201...
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JP Carballo
http://www.netfone2x.com
Bringing
these
or remove them then reinstall them? Sorry for my noobness as I'm
learning Linux for Asterisk.. I'm really a class 5 voice guy
tryin to
keep up!!
Thanks,
Kurt
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing
What's the voltage in the British West Indies?
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level,
I'm really quite busy.
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Asterisk-Users
a setup I have.
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JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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