Displaying 20 results from an estimated 100 matches similar to: "No subject"
2013 Sep 10
0
Looping an lapply linear regression function
Hi,
Try:
dat2<- read.csv("BOlValues.csv",header=TRUE,sep="\t",row.names=1)
dim(dat2)
#[1] 20 28
indx2<-expand.grid(names(dat2),names(dat2),stringsAsFactors=FALSE)
nrow(indx2)
#[1] 784
indx2New<- indx2[indx2[,1]!=indx2[,2],]
nrow(indx2New)
#[1] 756
res2<-sapply(seq_len(nrow(indx2New)),function(i) {x1<- indx2New[i,];
2011 Sep 02
0
No subject
crashing.
So, as a first step to solving **that** problem, make sure asterisk is
compiled with debug
flags, dumps another core file, and then you do the "gdb asterisk
<corefilename>", and
get a stack trace. That should give us some idea of what happened.
>
> I have a fairly simple Followme sequence in place to see how it works
> before I get into the complex scenarios.
2016 Mar 21
0
Networking in KVM
<div style="FONT-FAMILY: Arial; COLOR: rgb(0, 0, 0); FONT-SIZE: 12px"><div>Thanks all for the suggested tips. I confess I tried VMWare hypervisor esxi and found it less complicated to get set up and functioning correctly. <br /><br />I'll have to take up KVM another day, when I'm in less of a hurry to get something up and running right away. <br
2011 Apr 12
0
No subject
r>
<h2>Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010=
)
</h2>With SIP 3.2.X firmware (available on the Polycom download site)=20
and Asterisk 1.6.1, Polycom phones now support a full featured BLF=20
showing statuses of Ringing, Inuse and Online and one touch directed=20
call pickup.
<br>On the asterisk side all that needs to be done is to add a hint
2011 Apr 12
0
No subject
[0004f2xxxxxx](poly650)
defaultuser=0004f2xxxxxx
callerid="Front Desk" <1600>
mailbox=1600
*setvar=callidnum=1234561600*
and from extensions.conf:
[outgoing]
; Outbound unrestricted domestic calls
exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN}
on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.)
*exten =>
2011 Jan 10
0
No subject
Wait a second...
Action: DBGet\r\nFamily: DS\r\nKey: 0733025975\r\n\r\n
In the dialplan:
exten =3D> 0106024975,1,Set(DB(DS/0733025975)=3DINUSE)
exten =3D> 0106024975,n,Hangup()
exten =3D> 0106024976,1,Set(DB(DS/0733025975)=3DUNAVAILABLE)
exten =3D> 0106024976,n,Hangup()
Just a short call to my cell phone, to se if i get anything back, my =
cell phone doesn=E2=80=99t even ring.
Wait
2011 Sep 02
0
No subject
built-in; This doesn=92t matter because the moderator would have to use
meetmeadmin or the confbridge equivalent to control the other functions.
The moderator would either need two phones or a phone and a web =
interface.
Let=92s say Yves=92 =93special conference=94 is 5555. The moderator =
would start
using this command
Exten =3D> s,1,meetme(5555)
The participants would do
Exten =3D>
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
------=_NextPart_000_016C_01CBF83B.306A1A90
Content-Type: text/html;
charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2009 Jul 20
0
No subject
mailboxes).
Are you certain that removing either 612 or 610 mailbox would keep Asterisk
from complaining ?
>
> However, the MWI does not indicate voice mails for 610 and I keep seeing
> this error message:
>
> ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox
> 610 in context a10
>
> However, mailbox 610 is clearly defined in voicemail.conf:
>
2009 Jul 20
0
No subject
-- SIP/ vaso -e26c answered Zap/14-1
-- Executing DumpChan("SIP/ vaso -e26c", "") in new stack
-- Executing DumpChan("SIP/vaso-e26c", "") in new stack
Dumping Info For Channel: SIP/vaso-e26c:
============================================================================
====
Info:
Name= SIP/vaso-e26c
Type=
2011 Sep 02
0
No subject
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.
*same => n,Read(mobileNumber,app/input-mobile,10,,2,15)*
In the logs:
When it fails:
- - <SIP/ipbx-iwred-000002e> Playing 'app/input-mobile.slin' (language 'fr')
- - User disconnected
When it succeeds:
- - <SIP/ipbx-iwred-000002e> Playing
2009 Jul 20
0
No subject
in which Dial originally occurred, but for an unknown reason, it can't find
the appropriate hook to keep on.
Do you have any working sample ?
Regards
--0016e646050485a6cf0474456758
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Content-Transfer-Encoding: quoted-printable
Hello,<br><br>I'm using AEL2 (in Asterisk 1.6.1.6) and I can't find=
a way to successfully
2013 Apr 11
0
No subject
../libtool: line 1231: cygpath: command not found
You need to put cygpath in your PATH. This might also be why configure
is failing.
Best,
Tristan
--20cf307f35267e30eb0505f59648
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Content-Transfer-Encoding: quoted-printable
<div dir=3D"ltr">Hi,<br><div class=3D"gmail_extra"><br><div class=3D"gmail_=
2017 Jun 06
0
free Icecast windows source software with HE-AAC(/v2) support
2018 Mar 30
1
Tinc: performance
2014 Jan 08
2
Possible memory leak in Icecast-2.4-beta3
2020 Apr 22
0
Recommendations on intrusion prevention/detection?
<div dir='auto'>Iptables or ipfw you always can create tables / chains and feed those with desirable IP's to ban.<div dir="auto"><br></div><div dir="auto">Something like fail2ban does. Make a big list, remove one or other IP.</div><div dir="auto"><br></div><div dir="auto">On my setup, I
2020 Jul 05
2
Framebuffer double buffering (via FBIOPAN_DISPLAY)
<div dir='auto'>I am not familiar with that setting, but I have really struggled to find documentation on dealing with the framebuffer. Referring to this guide, "http://betteros.org/tut/graphics1.php#doublebuffer", I attempted to set the mmap allocation size to double, but it caused the mmap to fail. I no longer believe that it is a driver issue, though, because I just
2007 Jan 26
2
Hello Everybody, my problem with voicemail.conf
Hello everybody i am Ashish here.
i am new to this mailing list.
so dont know rules and regulation, just trying to post my problem of
voicemail.conf
Actuallt right now i am using Asterisk 1.2 on my LAN environment.
i am able to call all my extension very nicely.
Right now i am trying to deploying voicemail facility for all
extensions, so if anybody is not present, then he/she can leave
message,
2004 May 04
2
Can Asterisk support R2 signaling
Hi All:
I'm a newbee to Asterisk. I currently working on a project and want to know
if Asterisk does support R2 Signaling.
Thanks
Begra8fl
>From: asterisk-users-request@lists.digium.com
>Reply-To: asterisk-users@lists.digium.com
>To: asterisk-users@lists.digium.com
>Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs
>Date: Tue, 04 May 2004 13:32:00 -0500
>
>Send