similar to: test source for current xorcom rapid

Hi folks Xorcom Rapid 1.0 is avilable for download Q: 1.0? A: Sure, better than 0.9.1: * Asterisk 1.0.5 * Base packages upgraded * Built with SpanDSP support * Improved Zaptel detection * ast-cmd with some useful command-line abilities provided * ssh installed by default * putty.exe is included on the CD * music-on-hold files removed due to potential licensing issues

Hi all Xorcom Rapid 1.1 is here. * Asterisk 1.0.9 * Flash Operator Panel * improved Zaptel hardware detection: should hopefully detect E1, T1 ZapHFC and qozap. No more channel numbers guessing in zapata/zaptel.conf * and much of the "extra" software available for it You can get the full details at http://www.xorcom.com/rapid so I'll just highlight the points that I

Hi Version 0.9.0 of Xorcom Rapid Debian/GNU/Linux/Asterisk has just been released. Main changes: * A decent version of Asterisk/Zaptel (1.0.2) is provided * Includes a better default configuration * Automatic detection of the most common Zaptel cards * Contains more optional software (apache, MySQL, Webmin and others) - available via menu You can get it from:

Hi folks You are welcome to try our (Xorcom)'s latest debs (for Xorcom Rapid, or Debian Sarge in general) "Unstable": Asterisk and AMPortal: The repository is available at: deb http://rapid.dotsrc.org/ unstable/ #deb-src http://rapid.dotsrc.org/ unstable/ The commands you are looking for are: apt-get update apt-get dist-upgrade apt-get install amportal

Same question for asterisk-users as well: ----- Forwarded message from Tzafrir Cohen <tzafrir.cohen at xorcom.com> ----- Date: Wed, 7 Mar 2012 21:14:04 +0200 From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> To: asterisk-dev at lists.digium.com Short version: it's now time to remove. Anybody actually uses latest DAHDI with RHEL4? See inline, On Thu, Dec 29, 2011 at 07:42:39PM

Hi All, The past day or so I've setup a new * server based upon the Xorcom Rapid ISO. It did as promised; wiped the base system, installed Debian OS, installed Asterisk with a dummy configuration. So far so good. If I could get the config from my existing * server migrated to the Xorcom box the I'd be ready to roll. Essentially, it would be the same as I have now on Fedora Core 1, but

Hey Guys, Wanted a Suggestion..Howz this Xorcom Asterisk?I am using it and till now its fine as currently it is in testing stage with 3-4 users. Any Ideas??? Thanks Sharadindu Mohanty --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

Kapejod is working on a fix for the CDR problem in bristuff. See below ----- Forwarded message from kpj@junghanns.net ----- Resent-From: tzafrir.cohen@xorcom.com Resent-Date: Fri, 10 Feb 2006 13:01:25 +0200 Resent-Message-ID: <20060210110125.GU16880@xorcom.com> Resent-To: tzafrir@cohens.org.il Envelope-to: tzafrir.cohen@xorcom.com Delivery-date: Fri, 10 Feb 2006 05:19:50 -0500 Date: Fri,

static void senddialevent(struct ast_channel *src, struct ast_channel *dst) { manager_event(EVENT_FLAG_CALL, "Dial", "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name,

About the IVR, are u using Asterisk? Regards Bilal > ------------------------------ > > Message: 17 > Date: Wed, 18 Feb 2009 12:23:41 +0200 > From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> > Subject: Re: [asterisk-users] Credit Card processing > machines > To: asterisk-users at lists.digium.com > Message-ID: <20090218102341.GD21440 at xorcom.com> >

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP

---------- Forwarded message ---------- From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> Date: Oct 4, 2007 12:56 PM Subject: Re: [asterisk-dev] chan_h323 and chan_oh323 compatibilities To: asterisk-dev at lists.digium.com Hi On Thu, Oct 04, 2007 at 11:46:30AM -0300, Caciano Machado wrote: > I'm receiving a lot of warning messages from my Asterisk > 1.2.5/chan_oh323 every time

Okay, i'll move my discuss to asterisk-users. Thank you. On 1/17/07, Tzafrir Cohen <tzafrir.cohen@xorcom.com> wrote: > > > On Wed, Jan 17, 2007 at 04:39:03PM +0800, ??? wrote: > > Jonson Player wrote: > > > Hello, I intend to buy a FXO/FXS device from Linksys. > > > I'm thinking about SPA3102. What you guys thik about it. > > > Is ok, is

, address 2 Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.344611] ERR-xpp_usb: XBUS-00: x= usb_listen: usb_submit_urb failed: -19 Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.353325] INFO-xpp: XBUS-00: [] D= isconnecting Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.353373] Zaptel: Master changed = to XBUS-00/XPD-01 Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.353382] Zaptel: Master changed = to

After a reboot, I can't reproduce the problem anymore which is quite frustating. 2011/6/14 Tzafrir Cohen <tzafrir.cohen at xorcom.com> > On Tue, Jun 14, 2011 at 03:44:32PM +0200, Olivier wrote: > > Hi, > > > > I'm using a two-years old installation script for the first time on a > > Squeeze (linux 2.6.32) platform. > > For an unknown reason (might

one under my default context at extention.conf. And what is [pbx_config]? Thanks Eyal -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, June 25, 2010 4:05 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Is there a default dial plan that is not in

On Thu, Jul 15, 2010 at 10:19:10AM -0700, Steve Casto wrote: > > https://issues.asterisk.org/view.php?id=17529 > Thanks Tzafrir: > Unclear on how to apply patch, here is what I get: > [root at localhost asterisk-1.4.32]# patch -p1 < ../bug17529.diff.txt > can't find file to patch at input line 5 > Perhaps you used the wrong -p or --strip option? > The text

Thanks. The configure run successfully. but I got the warning below.. checking for the ability of -lsrtp to be linked in a shared object... no configure: WARNING: *** configure: WARNING: *** libsrtp could not be linked as a shared object. configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr configure: