similar to: asterisk-users Digest, Vol 55, Issue 64

Reason I ask is I may have a non-voice T-1 replacement project going on and I'm investigating my various options. Costs may be about the same for turn-key and DIY. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, February

Define non-Voice T1 porject? You do know that TDMoE does not travel over long distances, You can not route or otherwise take it off of a single ethernet segment. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Mike Hammett > Sent: Thursday, February 09, 2006 1:20 AM > To:

Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a full featured BLF showing statuses of Ringing, Inuse and Online and one touch directed call pickup. On the asterisk side all that needs to be done is to add a hint to the extension and enable directed pickup.

I don't know if 911 uses caller ID or BTN (Billing Telephone Number) 900 calls, operator calls, and 800 calls use the BTN not the Caller ID... Anyone???? 3. Re: E911 and asterisk (Martin Pycko) Message: 3 Date: Mon, 21 Jul 2003 12:05:38 -0500 (CDT) From: Martin Pycko <martinp@digium.com> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] E911 and asterisk

> -----Original Message----- > From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] > Sent: Thursday, March 16, 2006 8:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on > *HAandPolycomphone!! > > > > > > "Q: What are the plans for HA? > > That's BS. Last time I

Send CentOS-announce mailing list submissions to centos-announce at centos.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.centos.org/mailman/listinfo/centos-announce or, via email, send a message with subject or body 'help' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When

Struggling with an IP650 and 7 IP335s this morning. I have the following hints defined (courtesy of FreePBX 2.9): extensions_additional.conf:exten => 300,hint,SIP/300 extensions_additional.conf:exten => 301,hint,SIP/301 extensions_additional.conf:exten => 302,hint,SIP/302 extensions_additional.conf:exten => 303,hint,SIP/303 extensions_additional.conf:exten => 304,hint,SIP/304

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

I need to set the content to 0 because actually is not using malloc(), it uses calloc() that does set the memory to 0. Best Regards --------------------------------------------- Helios Cuevas L?pez Embedded Software Engineer Castle Care-tech Ltd. First Floor, 6 Bracknell Beeches, Old Bracknell Lane West, Bracknell, Berkshire, RG12 7BW Office - 01344 469470 Direct Dial - 01344 469485 FAX - 01344

es simple lo unico que tienes que hacer es instalar la libreria(xlsReadWrite) y direccionar y seguidamente especificar el numero de hoja de excel library(xlsReadWrite) dat.excel <- read.xls("d:/ruta del archivo/doc.xls", sheet=1) espero que te sirva ------------------------------------------ El día 18 de agosto de 2010 06:00, <r-help-es-request en r-project.org>

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

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

Hi John, Kevin, Tor and Wiley (and everyone else) - >> I guess the phone just doesn't register as busy when there is only one >> call on a line. It has to have two calls on a line appearance to >> register as busy. Has anyone figured out how to disable this hold >> feature and just have the second call go to the second line, the third >> call to the third line,

Hola a todos, Sigo la discusin sobre la web de usuarios R y me pregunto porque nadie ha mencionado "Ning". A m me parece una herramienta perfecta para crear las redes sociales, y estoy seguro que satisfacera mayora, si no todos los requisitos propuestos (foros, blogs, fotos, videos, miembros ...) Para crear uno : http://www.ning.com/main/createyourown Aqu podis ver un ejemplo:

> > >Noah, I could really use 1.3.4... however, a better question might be >how are *you* getting 1.3.4? I can't seem to get this from anyone, >including my reseller. > > I know how you feel! Polycom has made a product that sells like hotcakes, but they don't seem to want to support it at all, even when you do try to go through the proper reseller channels. I

Hi, I'm using the FLAC command line tool to encode my WAV files in some directory to FLAC files. Now I noticed sometimes the output of the tool is mostly like this: flac -V --best "14 The Hose - Time Warp (Old Skull Mix).wav" flac 1.1.4, Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007 Josh Coalson flac comes with ABSOLUTELY NO WARRANTY. This is free software, and you are

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Wednesday, August 22, 2007 10:51 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 37, Issue 88 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com

Funny how a topic will come up that you have never dealt with before, and suddenly it comes up from multiple directions at the same time. I was recently involved in a meeting where TAPI (which I understand only vaguely) was proposed as way to link a custom application to Asterisk for outbound and inbound call processing, much like SugarCRM and probably others are doing. Today I was asked by

On Thu, 2006-03-16 at 01:01 +0000, Guy Helmer <ghelmer@palisadesys.com> wrote: > David Wolfskill wrote: > > I'm running: > > > > localhost(6.1-P)[21] uname -a > > FreeBSD localhost 6.1-PRERELEASE FreeBSD 6.1-PRERELEASE #19: > Wed Mar 15 07:15:25 PST 2006 >