search for: intermediae

Hi, is geoRSS overlay supported with openstreetmap ? I can''t get it to work. Google maps georss http://blogs.intermedia.uib.no/motel/experiments/bergenGeoRSS.html Openstreetmap georss http://blogs.intermedia.uib.no/motel/experiments/bergenOSMGeoRSS.html As you can see, nothing shows in the openstreetmap example. Regards -- Rune Baggetun

Greetings - I have Fedora Core 2 Linux box with Samba 3.0.14a compiled and installed there. Here's output of testparm: <TESTPARM OUTPUT> [root@cflinux ~/Downloads/samba-3.0.14a]# /usr/bin/testparm Load smb config files from /usr/local/samba/lib/smb.conf Processing section "[homes]" Processing section "[printers]" Processing section "[tmp]" Processing

Hello friends, I have a script that uses the sftp client to transfer a file to another server using PK authentication. It was working until I upgraded from OpenSSH 2.5.1p1 to 2.5.2p2-1 yesterday (on a RH Linux 6.2 system). The sftp command is: sftp -o "IdentityFile ~/.ssh/id_dsa" \ $PUSERNAME@$PSERVER >$TEMPFILE 2>&1 <<-! cd $PDIR put $DOC_ARCHIVE ls quit !

Hi Thomas, It's a crazy idea if you are storing a few gig's, raid 5 would do, but we could be storing TBytes of audio and video, then using raid's or san's becomes more of a problem. That's why we are looking into using the cluster management of Oracle, but pull the audio data out and stream using Icecast instead of using intermedia. This solution abstracts us from the

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=498 cfilin@intermedia.net changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |chip@innovates.com -- Configure bugmail: https://bugzilla.netfilter.org/bugzilla/userprefs.cgi?tab=email ------- You are

Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to be able to "enhance existing IP-PBX solutions such as Asterisk", so maybe it can be used as a simple means to provide secure/encrypted calls. GNU SIP Witch - Summary <http://savannah.gnu.org/projects/sipwitch> GNU SIP Witch - GNU Telephony <http://www.gnutelephony.org/index.php/GNU_SIP_Witch>

...te) el martes 3 de Septiembre. Ven para descubrir cómo los datos se están convirtiendo en el nuevo petróleo de internet. Para las personas interesadas que estén en Madrid o puedan venir, nos reuniremos para aclarar dudas, dar más detalles, el próximo martes 27 de Agosto de 4 a 6, en el Terrario de intermediae http://intermediae.es/project/intermediae/page/terrario -matadero Legazpi. -- Pedro Concejero BigData - Analytics @ Telefónica I+D<http://www.tid.es/es/Paginas/default.aspx> Part of Telefónica Digital<http://www.telefonica.com/es/digital/html/home/home.shtml> / Telefónica Digital Hub...

hola a todos y todas, Estoy haciendo un estudio sobre las mareas y tengo dos variables que quiero relacionar, la altura del mar y el tipo de marea viva, muerta o intermedia. Unos datos simulados podrían ser: > datos <- data.frame(v1=sin(1:50), v2= rep(c("a","b","c"), each = 5, > len = 50)) Ahora mi pregunta, si dibujo la altura de la marea sería:

Asterisk users, I posted the following email to the Fedora users list <https://www.redhat.com/archives/fedora-list/2006-March/msg04154.html> and it got no responses, so now I'm calling on your expertise. Please take a look at it and share your knowledge on the subject with me. Additionally, let me know if you believe what I am trying to do is critically flawed. It's possible

I did not install soxmix in my linux box. If you having issues with mixmonitor, you can put both legs of the call into a conference and record the conference -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Matt Roth Sent: Thursday, April 13, 2006 1:20 PM To: Asterisk Users Mailing List - Non-Commercial

All, our dovecot installation provides a bundle of intermedia CA certificates using the ssl_ca option. 2.3.0 does not supply the bundle, resulting in various clients either complaining about an unverifiable server cert, or quietly not connecting. The log has Jan 5 17:01:46 Bounce dovecot: imap-login: Disconnected (no auth attempts in 0 secs): user=<>, rip=XXX, lip=YYY, TLS

Hello, We are about to attempt implementing realtime queues using Asterisk Business Edition and are wondering if anyone has had any experiences or recommendations /warnings with using this feature. One issue we're trying to decide on is if we should have a MySQL server running on the Asterisk server or a remote server to house the database/tables for the queue settings/members. The concerns

I just check the source code, Monitor uses ast_writestream and it eventurally goes down to au_write, g723_write, etc. They don't commit to the disk. So, in effect, if you have a lot of ram, the audio should stay in ram until it gets swap out or the file is closed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

List members, Is it possible to record outbound analog calls using an X100P? I was asked if I knew how to record all calls for a shop with 4 analog phones transparently to the end users. I thought Asterisk was a good fit for this and I envisioned using either Digium TDM400Ps or Sangoma A200s with 4 FXO and 4 FXS modules. The FXO modules would be connected to the existing PBX and the FXS

You got to be kidding about 53 calls being recorded at sametime is an issue. I have done at least twice as many on my dual xeon 3.4Ghz system and had no problem as clients like to record every call that goes through the system. Then again, in my system, the in and out channels are mixed first before they are written to the disk. ________________________________ From:

Except that mixmonitor still has a bug in it. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 12, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -sound quality-critical! Matt Roth

Just good old monitor with no mixing onto the scsi drive. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tamas Sent: Wednesday, April 12, 2006 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk - sound quality-critical! Hi, how do

Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=498 Summary: RTP packets are not hitting NAT table Product: netfilter/iptables Version: linux-2.6.x Platform: All OS/Version: Fedora Status: NEW Severity: major Priority: P2 Component: NAT AssignedTo: laforge@netfilter.org ReportedBy:

List users, Using Asterisk in an inbound call center environment has led us to pushing the limits of vertical scaling. In order to treat each caller fairly and to utilize our agents as efficiently as possible, it is desirable to configure each client as a single queue. As far as I know, Asterisk's queues cannot be distributed across servers, so the size of the largest queue we service