Displaying 20 results from an estimated 89 matches for "everts".
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events
2006 Nov 03
4
some simple newbie help with dialplan needed...
Hi all!
I need a simple plan for the following:
*answer call
*wait for 4 digit extension
*send call to 4-digit extension entered.
I tried the following, but that doesn't work...
exten => 998,1,Answer()
exten => 998,2,Background(agent-newlocation)
exten => 998,n,WaitExten(20)
exten => 998,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr)
WaitExten obviously does not fill EXTEN with
2004 Sep 22
1
'asterisk' displayed on my Cisco 7960 & 7912 ...
The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c
#define CALLERID_UNKNOWN "Asterisk"
I've changed mine to:
#define CALLERID_UNKNOWN "Unknown"
-----Original Message-----
From: Shaun Ewing [mailto:sewing@gmail.com]
Sent: 22 September 2004 14:16
To: Asterisk Users Mailing List
2004 Sep 14
1
What does 'Forbidden (From header is not a Trust host or gateway)' mean?
From a 'sip debug':
Sip read:
SIP/2.0 100 Trying
From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e
To: <sip:[dialled number]@[SIP server of VoIP provider]>
Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP]
CSeq: 102 INVITE
Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b
Content-Length:0
7 headers, 0 lines
Sip read:
SIP/2.0 403 Forbidden
2004 Sep 09
3
weird routing(?) problem with 2 Asterisk servers
Hi everyone!
situation:
Asterisk-server A: 192.168.11.6
Asterisk-server B: 192.168.2.44
server B contains a register => username:password@192.168.11.6
But... when I boot it, I get:
Registered to '192.168.11.6', who sees us as 10.138.3.2:4569
Why doesn't server A see server B as 192.168.2.44??
All other traffic going over these lines has no problems with this. The
2004 Sep 22
1
'asterisk' displayed on my Cisco 7960 & 7912...
Hi!
When I call a colleague of mine from my Cisco (via Asterisk), they get
on their display:
From Evert
asterisk
How do I remove/change the 'asterisk' part?
Regards,
Evert
2004 Sep 14
1
Wrong ID going out...
Hi all!
I'm trying to have asterisk route all outgoing calls out via my VOIP
provider.
exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in
the direct direction. However, debug shows that my asterisk doesn't
identify itself correctly:
Sip read:
SIP/2.0 100 Trying
From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa
To: <sip:[dialled
2009 May 07
3
RSPerl and Statistics::R
Greetings!
Being a Perl hacker for some time, and wanting to leverage what R provides, I've been trying to work with Statistics::R and RSPerl.
The former has a race condition that breeds some unreliability and the latter seems to have issues all around, and neither has been updated in some time.
Are these projects are abandoned, or is there some effort currently being undertaken to
2008 Aug 05
5
OpenSolaris+ZFS+RAIDZ+VirtualBox - ready for production systems?
Hi all,
I have been looking at various alternatives for a system that runs several Linux & Windows guests. So far my favorite choice would be OpenSolaris+ZFS+RAIDZ+VirtualBox. Is this combo ready to be a host for Linux & Windows guests? Or is it not 100% stable (yet)?
Greetings,
Evert
This message posted from opensolaris.org
2004 Sep 08
2
'connecting' voip-numbers to our Asterisk
Hi everyone!
I have a problem... We have received a couple of phone numbers for voip
from a local voip-provider. The work fine directly with a Cisco 7960,
but so far I've not been able yet to integrate them into Asterisk.
I've tried:
/etc/asterisk/extensions.conf
*****
[ip-incoming]
2004 Sep 17
2
dial '0' for outside line and get a dialtone...
Hi everyone!
I'd like to create the following: a user picks up the phone (gets a dial
tone), dials '0' for an 'outside' line, gets a second (different?)
dialtone, and is able to enter an external phone number.
How do I implement this in extensions.conf...?
Regards,
Evert
2010 Jan 18
10
Dahdi/callerid issue
Hi All,
Maybe someone knows this, im using dahdi in combination with a TDM400,
where 2 analog PSTN lines are connected.
The weird thing is tho that when someone calls the analog lines it goes
perfectly fine, the line comes in and all works ok.
Except:
Sometimes the callerid from the caller is not the complete number, but
only a few random numbers from that phonenumber, and sometimes its
complete.
2005 Mar 04
1
dialing from a website. How to start...?
Hi all!
We use a PHP-portal for management of our projects & contacts. Now I
would like to make it possible to dial contacts directly from the portal.
Since users have to log in, I can use that to determine which office
phone the call should originate from. And the number-to-be-dialed is of
course also listed.
How do I commence here? I'm pretty sure others have done this already,
so
2006 Jan 13
1
dnid support?
Hi all!
I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions:
913 - 11111 -> ext. 1
913 - 22222 -> ext. 2
913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider.
The config used here is based on Asterisk at home, so it includes also the
2007 Mar 01
2
DTMF not being detected with 1 provider. Works with the other provider...
Hi all!
Working on the following brain-scratcher. I am setting up a Trixbox
system for someone who uses 'provider A'. Everything works fine, except
for the IVR: keypresses by callers are not being detected.
Just for testing I added my own provider, 'provider B' to their system.
And then the IVR works!
Is there any possibility that the config on the provider-side is causing
this
2006 Jun 22
1
Trouble with windows mounts after reboot of windows server
Hi all!
Am I the only one with this problem? I doubt it...
The problem is that I have a couple of shares of a W2K server mounted with Samba on my (Gentoo) Linux. This works fine, until the W2K server gets rebooted. After that the shares are just timing out,
and they are impossible to unmount/remount... :-/
How do I prevent/fix this problem?
Regards,
Evert
2001 Dec 18
4
What systems are you using to listen to Oggs?
What rigs do you folks use to listen to your music? I have a P-III 500
with Altec Lansing speakers in the dining room and a P-II 350 with Labtec
speakers in the Guestroom/office. Sorry, I can't remember what model the
Lansings are off the top of my head. The Labtec speakers are fairly
cheap. I have a PCI ensonique sound card in the P-III system. I not sure
what kind of sound card is in the
2005 Dec 20
1
GE Digital Energy NetPro 19" UPS. Supported?
Hi all!
Does NUT offer any support for the GE Digital Energy NetPro 19" UPS? I have one of those beasts here, connected to a Linux machine via it's own 'serial' cable.
Regards,
Evert
2010 Sep 20
0
unz() ignores encoding argument
Hi!
I'm trying to read individual files from a ZIP archive, using the unz() function. Some of the files contain non-ASCII characters and I'd like to avoid unpacking them in a temporary directory.
My problem is that unz() seems to ignore the encoding="latin1" option I need to read the non-ASCII characters properly. I can't find a clear indication in the documentation that
2008 Dec 27
1
Zipf fitting using R
Dear R-users,
I am new to R and would like to use it for fitting the zipf distribution to
some numeric data that I have. Here's the snippet that I use:
library(VGAM)
X <- read.table(file("~\\mydata.txt", encoding="latin1"))
w <- as.vector(t((X[2])))
w <- w/sum(w)
y <- (1:length(w))
fit = vglm (y ~ 1, zipf, tra=TRUE, weight=w)
zipf(N=NULL,
2003 Dec 29
2
USER environment
Browsing in the 2003-September archives, I found out how to start dovecot
from inetd (using imap-login instead of imap). Still I've got a few
problems. I prefer not using inetd, but that seams impossible (see my 1st
email). And telnetting to port 143 results in an immediate connection close.
Hmmm, I'll continue exploring the archives.
Regards
Evert