search for: teleconference

?? 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 successfully and I'm able to participate as an "audio-only&qu...

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 requir...

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 audio mix of the conference is perform...

...pura SPA-1001 adapter - VoipJet for termination. - ulaw codec - Comcast cable modem connection Can someone help me narrow down where to begin to solve this? Should I try another termination provider? SPA-1001 settings? Different Asterisk settings? Should I provide my Asterisk conf files here? Is a teleconference service like freeconference.com more likely to have this type of issue than standard phone calls? Thanks. -- Lee

...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 knowing that the other end did not answer. 2. When calling an automated number such as AT&T Teleconference, I never get an indication of connection from the receiver, though I know they answered because I hear them. Any thoughts or suggestions. Danny Nicholas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments...

...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/2016/0...

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

...software was ViewCast's early Osprey, who's hardware consisted in large, and supplied through companies acquired by PolyCom. PolyCom's Hardware/Software solutions through acquisitions of many players at the time, were alot of the same stuff, their motivation of course, to produce early Teleconference products, and snap up major stakes in Streaming Media. One box was an encoding server that was used to encode the video/audio into digital format. The second was a server with a T1 internet connection and dedicated IP, the encoded video/audio files were stored. The third was a IIS webserver same T...

...software was ViewCast's early Osprey, who's hardware consisted in large, and supplied through companies acquired by PolyCom. PolyCom's Hardware/Software solutions through acquisitions of many players at the time, were alot of the same stuff, their motivation of course, to produce early Teleconference products, and snap up major stakes in Streaming Media. One box was an encoding server that was used to encode the video/audio into digital format. The second was a server with a T1 internet connection and dedicated IP, the encoded video/audio files were stored. The third was a IIS webserver same T...

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:

...dle buggy upgrades if I can access different firmwares for free, but to pay for a firmware that won't even load? No thanks. The speakerphone is by far the best speakerphone I have ever had on any phone, ever. I use this phone when I do training conferences and meetings that are broadcast via teleconference, and it performs like a champion. The phone has good looks and is definitely an eye catcher. I have a couple of them set up in our administrative offices as "eye-candy". The later SIP versions work well. ATA: Sipura SPA-2000 Good: Configuration, Functionality, Stability Bad: Unimpress...

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