similar to: Asterisk and Tandberg VCS

Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's

Has anyone had any luck getting Asterisk 1.6.2.13 to register to a Tandberg Gatekeeper? The logs on the Asterisk end seem to show that the registration request is sent, and the Tandberg Gatekeeper responds. However, the response doesn't seem to be what Asterisk was expecting. Here is my ooh323.conf, followed by the relevant portion of the h323_log: [general] port = 1720 bindaddr =

> -----Original Message----- > From: centos-bounces at centos.org [mailto:centos-bounces at centos.org] On > Behalf Of Johnny Hughes > Sent: Thursday, May 19, 2005 8:48 AM > To: CentOS ML > Subject: Re: [CentOS] web.vcs.u52.k12.me.us/linux/smbldap problems > > On Thu, 2005-05-19 at 09:41 -0400, Robert Moskowitz wrote: > > I am perhaps going about this the hard way of

For the last 10 years we have had some Tandberg video conferencing units spread across our WAN. To make a long story short we had to work with another entity to allow access into those units with H.323. Calls from public networks have never worked to those devices with private IP's. Over the summer, my coworker and I built an * VOIP system for one of our high schools. The need came up for

Hi, I currently have some systems on AsteriskNOW 1.7 & have been happy with its clean simplicity & reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick with their previous philosophy of keeping everything very vanilla & making it clean & simple for someone who understands how to manage CentOS, FreePBX, tftp, ntpd, etc. but

Hello, Is it possible to set things such as parts of config files are edited though AsteriskNOW GUI while other parts remain "hand editable" ? AsteriskNOW website include screenshots but not much information (such as user manual) beside that. This thing was the one that kept us from using freepbx (let me say I don't mean it's not possible with freepbx : I mean we couldn't

Hi, I've a problem configuring my Asterisk. What I try to reach is to interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP) with 1 constraint I can't change : "every RTP flow needs to pass THROUGH Asterisk, and are NOT nated" What I observe : - a call made from a SIP Phone registred in Asterisk to Tandberg works (voice and video bidirectionnal) - a call

Hello all. As part of a project I'm working on to migrate from Asterisk 1.0.11.1 to the latest LTS version, I'm looking into providing a HA/failover solution for the new Asterisk installation I'll be deploying. It would appear my best bet would be to use the R850 Digium appliance. Does anyone have any experience running one of these devices along with AsteriskNOW? I ask because

I am currently using AsteriskNow v2. What I am trying to accomplish is having all calls from an area code go directly to the person responsible for that area. While searching for a solution for this I did come across a post that had a few examples. So Josh at extension 1902 would receive all calls from the 808 area code. exten => s,1,GotoIf($${CALLERIDNUM:0:3}" = "808?1902|1)

Hello all. I was hoping someone out there might have some advice or suggestions regarding an upgrade from an archaic Asterisk version. I've been given the daunting task of upgrading a very old Asterisk-1.0.x install to a recent LTS version. I'll also need the install to have high-availability and failover support. From my research, it would appear that AsteriskNOW-3.0 might be my

I received this email link from Macromedia today http://www.macromedia.com/newsletters/edge/february2005/index.html?secti onIndex=3&trackingid=AWQX They are now selling Breeze video conferencing server by the minute as an asp, this service is powered by the macromedia communications server which is just a sip server like asterisk but for video. I'm still offering a $2,000 bounty

We have here a TandBerg "videoconferencing system" connected in asterisk with a Beronet card BN4S0 (4 BRI ports). I`m trying to make a videoconferece (video + audio) with this Tandberg to another Tandberg using the ISDN channels through the BN4S0 BRI Card. But, i'm only obtaining audio calls, the video not appears. When i try to make a 64K calls (data calls), the call does not

I've moved from trixbox to AsteriskNow. Does anyone know if there's something like the PBX Status screen for AsteriskNow? A module the shows the status of SIP and IAX2 registry and peers, etc for all individual entries? The FreePBX System Status screen shows when something fails to register but doesn't show who it is. Frank

Anyone know what they are going to be announcing? Any relevance to Asterisk? When are we going to see a decent video conferencing implementation into Asterisk? 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: Polycom Connect

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

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

Hi, ? sip.conf is missing in /etc/asterisk after installing the package AsteriskNow 1.5 Beta release? Is there any guide for FreePbx Administration? Thanks ? Explore and discover exciting holidays and getaways with Yahoo! India Travel http://in.travel.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey Guys, I have installed AsteriskNOW and I've found that the SIP module is not installed (along with all the SIP configuration files) until I log into the freepbx webGUI and click the reload button. Is there a way to get all the sip configuration files and module installed without having to do a reload from the webGUI? -------------- next part -------------- An HTML attachment was

Recently, I've been working on tools for doing high-quality conversions of project histories from centralized version-control systems like Subversion to modern decentralized systems like git and hg. To see what I mean by "high quality", take a look at my DVCS migration guide: <http://www.catb.org/esr/dvcs-migration-guide.html>. A really good conversion should, for example,