similar to: video conference feature

Hi, I have installed latest stable version of Asterisk. I registered 1 Quintum Gateway and 5 eyeBeam SoftPhone. But I can make just one call at the same time, if i try to make calls from 2 softphones to anotherone, second caller listens " the person have extension xxxx is on the phone ...." . So we couldn't make two or more calls at the same time for a SoftPhone. What should we

Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used "canreinvite=yes" but it didn't work. Description from asterisk conf. File; (canreinvite=yes ; allow RTP voice traffic to bypass Asterisk) Thanks Erdem HAKI - erdemh@tesas.com -------------- next part

Hi list, I set up two asterisk servers , 1001 is the first asterisk server's sip user, and 2001 is the second asterisk server's sip user. Each of them work well, but I don't konw how to connect them. I want to let the user in 1th Asterisk can call the user in 2nd Asterisk. First asterisk server ip : 192.168.3.101 Second asterisk server ip : 192.168.3.102 can someone

Hello Haki I fixed this problem following the instructions in /usr/src/zaptel-1.0.9/README.udev. Regards Cec?lia -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Erdem HAKI Enviada em: sexta-feira, 15 de julho de 2005 05:11 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto:

I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first number's call to Trunk if second number is unregistered. Is it possible? ?f yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567---------------------------- registered to Asterisk

Hello, I'm trying Meet Me Feature. I read wiki , searched google and i configured my extension.conf and meetme.conf. But I receive "this is not a valid conference number, please try again" message, so what could be the problem? Thanks for your interest. Erdem HAKI -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there

--- listas iPfone <listas@ipfone.com.br> wrote: > Hi all > > I need to know if the video support for h.263 is > active in version stable > 1.0.7 to use with eyeBeam in asterisk it works for me... [2222] type=friend secret=xxxx auth=md5 callerid="myCallerId" <2222> canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex

Sorry if this shows twice but it appears my first message was quarantined because of my digital signature. All, I have been tasked with setting up video conferencing utilizing asterisk. One of the requirements is a softset that has video capabilities. Eyebeam looks promising but I was just wondering if anyone out there knows of any freeware with comparable features of Eyebeam that they

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

Hi all of you. I am trying to configure voice mail in asterisk and i am facing problems. I have found following warning message in /var/log/asterisk/messages -------------- No application 'Voicemail' for extension (macro-mainmenu, s, 5) I have configured voice mail accordingly in extention.conf [headoffice] -- ------------ ------------- exten => _63,1,Macro(mainmenu)

Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start

Hi Erdem, Since it is a target specific pass, you should put your pass inside the target's folder (e.g. lib/Target/X86/) and declare your pass in Target.h. Then you need to modify the target's TargetPassConfig ( http://llvm.org/docs/doxygen/html/classllvm_1_1TargetPassConfig.html) to enable your pass. You can check out the existing MachineFunction passes (e.g. X86ExpandPseudo.cpp) to get

Hi All, Got some problem with asterisk video, i'm testing an eyebeam and a grandstream video phone. call from grandstream to eyebeam works ok, video shows up. but calls from eyebeam to grandstream there's no video, but audio is ok. Regards, Nhadie

This seems to be what I am looking for. That was very helpful. Thank you. Erdem From: vlknkls at gmail.com Date: Wed, 16 Sep 2015 16:25:29 +0000 Subject: Re: [llvm-dev] How to add NOP? To: erdemderebasoglu at hotmail.com; llvm-dev at lists.llvm.org Use MachineInstr::memoperands() function to get memory operands then you can get the address space by using MachineMemOperand::getAddrSpace(). Volkan

Can you send the IR you are using? Volkan On Thu, Oct 8, 2015 at 6:28 AM Erdem Derebaşoğlu < erdemderebasoglu at hotmail.com> wrote: > Thanks. I enabled my pass. I have one resolved issue though: > MachineMemOperand::getAddrSpace() always returns zero. How can I use it to > distinguish private memory accesses? > > Erdem > > ------------------------------ > From:

I am using eyebeam and I am happy with it. However, it is boring just to talk to my son in the other room. Whenever I try to convince somebody to buy eyebeam, they are scared of the price. Is there a free video soft phone available, that will work with eyebeam / asterisk? bye Ronald Wiplinger

I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ;