search for: teleconferences

?? Hello.? We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).? We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.? If a person starts talking they will cut off others on the call.? Is this normal behavior?? Are there any options I can change to change this? ?? Thanks! James -------------- next part -------------- An HTML attachment was scrubbed......

Hello list! I am new in * but i want to learn about its possibilities. I want somebody to tell me if what I want to do is possible with *. I have a teleconference tool which uses SIP and now I am using Asterisk as POTS gateway. When I dial certain number from a telephone I connect with asterisk which asks me for an extension. When I dial certain extension I connect with my SIP application

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hi Speex devs! This is firt a warm thanks for your efforts and for making Speex a reality. Well done! I developped a teleconference application, using Speex as the codec for the audio layer. The project is http://www.encours.org. In the course of this project I had the following use case: - Participants connect to the conference. Each participant sends a Speex encoded stream. - A common

I've been having audio breakup problems (on my end) in my Asterisk tests. I'm not sure of the most likely source of this quality problem. 99% of my LD calls are calling into a tele-conference service called freeconference.com for group meetings. Its a free phone conference system that works quite well with pstn phones. I've been using it for quite some time. But the audio problems

Greetings listers, I've posted this at least once previously, but thought I'd try again. I've got a TDM410P card on Asterisk 1.4.21.2 and experience these two problems. 1. When placing an outgoing call, I get no audio until Asterisk bridges the connection (2-15 second delay). I can Answer before Dialing, but this give me a incorrect CDR and no way of

...c design principles behind the new API? how can adoption of such an interface work? Please let me know whether you?re interested in this - we would be glad if some of you would join this effort and participate in the working group. The working group will operate via email [2], a wiki page [3] and teleconferences. We also plan to have an in-person meeting at this year?s useR! conference. Also, if you have someone in mind that should be part of the working group, please let us know. We may not have thought of them yet. Best Regards, Lukas Stadler [1] https://www.r-consortium.org/news/announcement/2016/03...

Hello All, I am looking at doing some video conferencing with SIP. I was hoping to get some early pointers from any one that is currently doing this. I have been all over goggle and voip-info and there is a ton of anecdotal information but, I was hoping for more specifics of what people are actually using that works and even some of what hasn't worked so that I can stay away. What I am

Hello all reading this. I am new to the list as a way to learn a few things about IceCast and I am fuzzy on. I am not a total noob to media, or serving it. I have even developed, setup configured, a media server, or two, and many many dedicated, or otherwise hosting servers, and by either blind luck and chance found them to actually work once I was done. I am Scott, am in the process of building

Hello all reading this. I am new to the list as a way to learn a few things about IceCast and I am fuzzy on. I am not a total noob to media, or serving it. I have even developed, setup configured, a media server, or two, and many many dedicated, or otherwise hosting servers, and by either blind luck and chance found them to actually work once I was done. I am Scott, am in the process of building

Hi all, To solve my need for dial in modems, I've hit upon an idea: buy a used T1 "analog" modem bank like a Lucent Portmaster that takes in a T1 and provides several 56K modems. This is overkill for a lightly used dial in service, but the prices of these boxes is so cheap (~$300) with ISPs going away from dial in service that it makes sense. This is how I imagine it would work:

Good evening. I just wanted to take a minute and review my experiences with some of the SIP devices out there on the market. I hope this post will help newbies or someone considering a certain device. I would appreciate any other input on either the devices I am "reviewing" or other devices that I didn't! These devices are deployed in our primary line and small PBX replacement

The disaster in the Gulf coast and the less than optimal initial response suggests to me that citizens must shoulder more responsibility for emergency management. Communications loss must have played a large role in the failures that occurred. I can't help but wonder if there are fewer ham radio operators today and that if there were more, maybe they could make a difference in future

Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So