search for: gilroy

Displaying 17 results from an estimated 17 matches for "gilroy".

2004 Jul 02
3
CDR shows billsec=12 for all bridged calles.
...watching the Master file while making a call I see it updated at 12 seconds even while im still 'in' the DIAL app and the call continues on just fine. Iv looked through all my scripts and cant see anything to cause this. Im pretty new to asterisk so I don't know what to do now.. Morgan Gilroy Support i2 Networks Ltd tel 0871 717 7540 fax 0871 717 7541
2005 Sep 27
10
Software only Asterisk PBX (commercial)
Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker
2005 Feb 21
1
setting caller id number and using sip type=peer for incomming calles.
...hat username/secret against its list of users. if it still doesn't find it then drop it into the guest account. iv posted a bug with a bit more detail but it was closed as a configuration issue (which i suppose it is...) http://bugs.digium.com/bug_view_page.php?bug_id=0003621 Morgan Gilroy, Telappliant Support -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050221/ad8a36b9/attachment.htm
2004 Aug 19
2
Dial from AGI [MSG]
Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'
2005 Sep 21
0
Intermitant delays on call setup.
...both phones. It does this 'bunching' a couple of times then settles down to normal. Id say it was some sort of timing problem or load problem, but during these times conferencing etc works ok and there is no appreciable load on the server or network. Anyone have any ideas? Thanks. Morgan Gilroy.
2005 Sep 27
1
[MSG]TDM Error on ASUS Pundit-R
...ve looked around but I cant find an answer for this, I randomly get the error 'TDM PCI Master abort' and the system locks up. All I have found so far are a couple other posts on it but no solution. Running fedora core 3, asterisk stable, zaptel stable. Any help will be appreciated. Morgan Gilroy
2002 Feb 22
1
Wine scipt won't start
Is this a version that you compiled? If so I would suggest re-compiling it with the new kernel Cattab Cattabiani wrote: > That's my problem: i've recompiled the kernel > and the wine script doesn't run nomore. The error message is: > "/proc/sys/fs/binfmt_misc/register: > no such file or directory." > The directory is there but i can't write in that
2002 Feb 23
1
A Previous Setup Has Not Completed
I keep getting the error that a previous setup/install has not completed, please restart your computer and try again whenever I try to install new software. I have cleared all of my temp files but still get the error...anyone know how to fix this?
2006 Feb 08
2
SV: GotoIf number exists in file. How can i do this?
...HP, how do I make it output to an Asterisk variabel? I need to set a variable in asterisk to TRUE or FALSE based on the result of the PHP-script. ________________________________ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Morgan Gilroy Sendt: 8. februar 2006 15:28 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] GotoIf number exists in file. How can i do this? It will probably be easier to write an AGI script to do this, I cant think of anything in the dialplan to do this. -----Origi...
2006 Feb 08
2
Performance differences 64-bit vs 32-bit
Hi Guys. We've got a new server that we're looking to use for an Asterisk install. The CPU is a 64-bit AMD Opteron 246, 2Gb RAM. We're having some compilation issues with some of the Asterisk modules using Debian-amd64, and due to time constraints we are considering going 32-bit for now just to get the box up and running. Does anyone have any ballpark figures for the performance
2006 Oct 18
0
R issue with quantile using its package
...r the reply so what I did in the interim was unistalled Hmisc and chron and used version 3.0-2 of Hmisc which doesn't have the dependency on chron, and reinstalled its version 1.1.4 However I still heave the issue, when I try to run the quantile command on the given dataset. Thanks, ~Lloyd Gilroy, Lloyd (GTI) wrote: > I currently have an instance of R running on Solaris 8 Version 2.2.0 > (2005-10-06 r35749) > > Package: its Version: 1.0.9 > > Package: Hmisc Version: 3.0-1 > > > > I decided to build a new install on Linux RedHat As4.0 32 Bit running R &gt...
2005 Feb 21
1
setting caller id number and using sip type=peerfor incomming calles.
> > To get around this i updated CVS HEAD and changed the sip entity from > > type=user to type=peer (yes peer!) (type=friend works too but im making > > a point) the client now must register to set his outbound caller*ID > Number. > > Yes, that is normal. SIP has difficulty separating the remote party > identification from the authentication identification
2006 Feb 08
4
GotoIf number exists in file. How can i do this?
Hi there. I currently have a GotoIf statement that goes to a special extension priority if the CID match with one of the numbers in my "list" of CIDs. The way I've done it now is by multiple OR operators. There must be a better way. Anyone got some suggestions? This is basicly what I want. "If CID Exists in $File, goto s,10". So when I want to add a new CID I
2007 May 22
3
Dial out issues.
Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work
2006 Oct 18
0
(no subject)
I currently have an instance of R running on Solaris 8 Version 2.2.0 (2005-10-06 r35749) Package: its Version: 1.0.9 Package: Hmisc Version: 3.0-1 I decided to build a new install on Linux RedHat As4.0 32 Bit running R version 2.4.0 (2006-10-03) Package: its Version: 1.1.4 Package: Hmisc Version: 3.1-1 Package: chron Version: 2.3-8 I am using the library(its) on both machines and
2005 Feb 22
0
setting caller id number and usingsip type=peerfor incomming calles.
> Yes, exactly (and there will be other settings as well, to identify the > type of peer (network, trunk, endpoint) for other reasons). > cool, I really should read the lists more :) > > That's coming too, but in a different way. Actually if your remote peer > can send you Remote-Party-ID headers now, you can set "trustrpid=yes" in > your peer definition
2006 Feb 10
0
Any way to grep through fast moving consolemessages?
Yeah I do this, 1. create 2 ssh sessions to the same box, 2. on the first session do `script -f /tmp/astcli` 3. `asterisk -r` (and whatever other options you need 4. on the second session `tail -f /tmp/astcli | grep -i 'bob'` (on the grep you may have to ignore control chars if you have colour at cli, I think that's the -a option) then you can modify the grep to look for any