similar to: OT: HUD3 and NON-Trixbox Asterisk?

Hmm beginning of the end of free trixbox by the sounds of it. It was good while it lasted but time to download the latest iso while it's still available by the sounds of it. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ________________________________ From: trixbox

I thought I could try to post this question here since I have at times found excellent support here on A@H issues as well as pure/straight digium/asterisk issues. I am looking to finally build a permanent asterisk server and would prefer, if at all possible, to stick with the "new" A@H called trixbox. Does anyone know if the CentOS behind trixbox is a relatively complete CentOS system? I

A set of scripts were recently discovered in the trixbox line of PBX products, which connect to a remote host every 24 hours, to retrieve an arbitrary list of commands to be executed locally. These scripts were added under the guise of submitting 'anonymous usage statistics', however, with the help of DNS pollution, or malice on the part of the sponsoring company (Fonality), all

Looks like it's time to fork...... > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Lenz > Sent: Monday, August 13, 2007 7:28 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] FW: The trixbox Revolution Continues! Sign >

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read

I just read on Slashdot (at http://yro.slashdot.org/article.pl?sid=07/12/16/222243 ) that Trixbox "has been phoning home with statistics about their installations", as a Trixbox user exposed in "Trixbox Phones Home" at http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home . -- (C) Matthew Rubenstein

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

> -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Saturday, June 17, 2006 2:30 PM > To: asterisk-users@lists.digium.com; Douglas Garstang > Subject: Voicemail with NFS (working, I think) > > I'm using a stand-alone VM server and exporting the VM files ro for > MWI function only. All my registration servers mount the remote

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

-----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

Hi, This seems to be a fairly common question, but I have Googled for this quite a bit and looked at the Asterisk documentation/book and haven't been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and

I am interested in IMAP Voicemail storage for some of my customers. Does anyone know of any vendors of asterisk appliances (physical PBXs) that provide this as a "standard feature" (or an optional standard feature)? Ultimately, I'd like to be able to have a single point of accountability for the system as a whole. I would like an intuitive & powerful configuration GUI (such as

I love TrixBox, with the custom config files you can tweak pretty much with TrixBox too, I have at least done some. Plan to do a plain Asterisk install later, but for now I learn a lot about the config files just with TrixBox. Some things might be a bit harder with TrixBox due to some of the premade dial plans, but can get it to work :-) _____ Fra: asterisk-users-bounces@lists.digium.com

*Good morning! * *I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (it colors show channeltypes)ele no demostra the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the

Is this with Asterisk in the RTP stream? Is it doing any transcoding? > -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Tuesday, April 18, 2006 9:34 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Asterisk Performance 350 Concurrent > ChannelsWorking Nicely > > > Hi All, > > This is a performance

OK this was an easy one to fix. All I had to do is RTFM. Note on the wiki: ATTENTION: Make sure you take a look at bug report 7144 Just do what Kevin said, include the regcontext in whatever static context you have the priority 2 extension and don't make a static regcontext in extension.conf. Let sip module do the rest. Works great. Thanks Guys. JR On 12/5/06, JR Richardson

On 1/28/08, JR Richardson <jmr.richardson at gmail.com> wrote: > > You need to take a step back and first test the script without using > > MRTG. Execute it like this: > > # /opt/bin/asterisk-mrtg -h localhost -u XXX -p XXXX -1 SIP -2 Zap > > 10 > > 10 > > 10 > > 10 > > > > You should get 4 lines of numbers. That respresents your SIP

Hi list, I recently installed the Trixbox distribution as an HVM and have now converted it to a PVM. Found very few resources on Internet where people had done this conversion so I had very little to compare with. Wanted to ask for comments about the kernel used in the PVM after the conversion. Installing Trixbox and later installing the xenified kernel, gives a kernel of 2.6.18-92..... In my

Hey, I'm hoping somebody knows the answer to this. The script works fine on the old Trixbox 1.0 but have recently upgraded (just testing in VMWare) to Trixbox 2.2 What happens is Trixbox will drop the call after I call the AGI command in my dial plan. I first of generate a call file to call the user, then connect them to an extension in the dial plan [voice-report] exten =>

Just in case anyone else needs to know - there's an exploit 'in the wild' for Trixbox (which is CentOS based) that allows malicious code to be installed on a server. I discovered that one of my Trixbox servers was running 3 instances of a perl-based IRC botnet process called httpdse and this was pegging the CPU at 100%. Notes, comments, removal instructions, patches etc. here: