Displaying 20 results from an estimated 30000 matches similar to: "No subject"
2009 Aug 29
0
asterisk-users Digest, Vol 61, Issue 84
On Sat, Aug 29, 2009 at 10:30 PM,
<asterisk-users-request at lists.digium.com>wrote:
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2008 Jan 17
1
asterisk-users Digest, Vol 42, Issue 51
hi all,
how to set the caller id facility for
the TDM400p card.
Please help me
thanks,
sandeep.s
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From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, January 15, 2008 3:09 PM
Subject: asterisk-users Digest, Vol 42, Issue 51
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2011 Apr 12
0
No subject
system() to execute this script since it is (obviously) not really an AGI.
I'm guessing that system() would be slightly more efficient than agi().
Both require a process creation, but agi() requires (slightly) more
Asterisk resources in setting up the AGI environment.
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards
2009 Aug 31
0
asterisk-users Digest, Vol 61, Issue 85
Topic 6: RE:unable to execute command
hi there
i tried to execute the command as suggest like
exten => 1987,1,Playback(posix-restarting)
exten => 1987,2,wait(1)
exten => 1987,3,System(/usr/bin/python /home/docas/Desktop/mess1.py)
exten=> 1987,4,Hangup
it still doesn't work,now it does it give unable to execute command but
it doesn't reach the system command it just
2009 Apr 29
1
Bounty for parking on <slot>@<context>
Wrong list. asterisk-dev is for changing the C source code of Asterisk. I
don't think AGI's "count" or are considered for inclusion into the
subversion repository as stated by one of your conditions for payment.
On Wed, 29 Apr 2009, Alistair Cunningham wrote:
> I'd like to offer a bounty for a feature for Asterisk where an AGI
> program can park and retrieve calls
2011 Sep 02
0
No subject
core show function SIP<TAB>
I use:
set(PEERIP=${SIPCHANINFO(peerip)})
in one of my dialplans. For AGI, whatever function in your library that
executes 'GET FULL VARIABLE' should do the trick.
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline
2014 Nov 18
2
AGI and AMI in PHP -- What's current?
I'm writing some code that needs to access AMI in PHP. (I'll probably be
doing AGI later as well.)
I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and
appears to be a bit behind current Asterisk -- No event handler for event
'fullybooted'.
What PHP framework/library are you using -- and why?
--
Thanks in advance,
2011 Sep 02
0
No subject
is typing his number, though there is a 15 seconds timeout, and even if =
I type the number very fast it still may happen to me.<o:p></o:p></p><p =
class=3DMsoNormal><o:p> </o:p></p></div><p class=3DMsoNormal>It has =
been my casual observation that the speed at which I enter digits on my =
phone is unrelated to the speed at which my
2009 Apr 29
0
Verifone-Asterisk-AGI
Wrong list. asterisk-dev is for changing the C source code of Asterisk.
That's part of why you didn't get a response yesterday.
On Wed, 29 Apr 2009, Juan Miguel Quiros Arrieta wrote:
> I have to develop an application using the VeriFone vx510 device and I
> read this device needed or could use a PPPoE connection in order to
> validate and send all information collected from
2009 Jul 08
0
[asterisk-user] AGI control stream file
Trying to redirect to -user...
On Tue, 7 Jul 2009, Bryant Zimmerman wrote:
> Hey guys I posted this earlier and did not get any responses.
You posted what appear[s|ed] to be a user question to the dev list.
I did reply (on June 3), but I may have mis-understood.
> I am working on some AGI development that requires control of audio file
> playback. The control stream file is working
2009 Aug 08
0
DeadAgi application not exiting
On Sat, 8 Aug 2009, Max Alex wrote:
> Actually the scripts which are set to run to the hangup of channels,
> which is originated for sending fax. We are trying to get the answer
> time, duration of fax on hangup of that channels, but the script becomes
> stuck and we need to restart the asterisk and also we are not getting
> any output of script as it is stuck.
Let's start
2014 Aug 22
1
Can't hangup channel from CLI
Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting
Asterisk from a Tekelec T9000.
I'm accumulating stuck channels.
I'm googling now and I recognize that Friday afternoons are the worst time
to ask questions, but I'm getting desperate because this is keeping me
from rolling a system out to production. (Yup, I know. Who rolls out a
system on a Friday
2009 May 07
1
How to get meetme participants in dialplan?
The meetmeadmin() dialplan function lets you specify a user to mute,
un-mute or kick. But how do you get a list of users in your dialplan?
When a user joins a conference, the user number assigned is "the last user
number +1." If you have a long running conference with callers joining and
leaving all the time, this can grow to be a large number.
I want to be able to
2005 Mar 07
0
iax2 setvars help needed
I'm trying to pass a variable between servers using "setvar" in iax.conf.
I have a box (ts2) with a t100p in it. It answers the call and dials
another box (ast0) via IAX. I want to pass a variable along with the call
from ts2 to ast0.
I'm running CVS-HEAD-03/07/05 on ts2 and ast0.
ts2's iax.conf:
[general]
disallow = all
allow
2009 Jul 10
0
Meetme problem (talk detection/opt) in 1.6.1.1
On Fri, 10 Jul 2009, Jared Mauch wrote:
> I need the 'talking' information to better identify rogue people
> on bridges.
I'm a 1.2 Luddite so I don't have all these fancy new features :)
A different solution to a similar problem.
I had problems with abusive callers in my conferences. I whipped up some
dialplan and AGI mojo to let an admin mute and unmute individual
2014 Mar 21
3
Need more meetme users -- hitting some limit
I'm trying to determine the capacity of my host running Asterisk 11.8.1 on
CentOS 6.5.
The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet.
The primary application will be bridging groups of users using meetme().
I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1
behaving a bit more like a production box -- bridging calls (box2).
The call
2011 Apr 12
0
No subject
lob1
[lob] Show IPA
verb, lobbed, lob=C2=B7bing, noun
=E2=80=93verb (used with object)
1.
Tennis . to hit (a ball) in a high arc to the back of the opponent's=20
court.
2.
to fire (a missile, as a shell) in a high trajectory so that it drops ont=
o=20
a target.
3.
Cricket . to bowl (the ball) with a slow underhand motion.
Who do suggest we should be lobbing our fax machines at?
--=20
Thanks
2009 Jul 20
0
No subject
Look at the [/usr/sbin/]safe_asterisk script. It didn't serve my
purposes, but "inspired" me to write a script that did.
Basically, you create a process that creates the process that runs
Asterisk so it can do something if Asterisk exits.
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com
2009 Jul 20
0
No subject
echo test | mail -s test thomas.perron at gmail.com
If that doesn't work and you don't get any useful clues from the command
output, start digging where your syslogd logs messages.
Next, from a shell command line, try
echo test | mail -s test 5555551212 at txt.att.net
Note that this recipient is specific to this carrier.
If that works, it should work in Asterisk assuming you
2009 Jul 20
0
No subject
depends on where you are in the world. Generally speaking, somewhere
around 8 to 12.
There are many advantages to PRI over POTS:
) "Instantaneous" call setup.
) Higher reliability.
) Less cabling behind the server.
) Better support from your provider.
) More features.
) Better audio quality.
IMO, you should always use a PRI unless you can't afford it.
--
Thanks in advance,