search for: app_conference

Do anybody could compile app_conference on x86_64??? I tryied with two versions of app_conference and got the same problem on compiling: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared recompile with -fPIC app_conference.o: could not read symbols: Bad value" ENVIRONMENT: --------------...

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)? Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted...

Any C gurus out there that can tell me if this code compiled ok to be used in x86_64 (Pentium Dual Core). It's for the app_conference application. Im using Centos 4.3 x86_64 kernel: 2.6.9-34.ELsmp libgcc-3.4.5-2 gcc-3.4.5-2 after the compilation part is the makefile ************begin compilation******************* [root@centos app_conference]# make clean rm -f *.so *.o app_conference.o conference.o member.o frame.o cli.o libspe...

...nning in Xen virtual machines. Unfortunately, this kind of virtualization makes a real time clock impossible, which in turn makes ztdummy or a Zaptel driver impossible to load, which also makes MeetMe conferences impossible. As an alternative, I have downloaded, patched, compiled and installed the app_conference source code against the headers in Asterisk CVS HEAD. I can load the module into Asterisk and even connect to a conference channel with two phones. But each phone cannot send audio to each other. They just connect and go silent. Is this the current state of app_conferences development? I have rea...

Hi all, I've been trying to get meetme working for a while now (complie problems - will probably try again later on another machine) but have given up and started looking at alternatives. I've managed to get app_conference compiled and installed - show modules shows its there in asterisk, but I don't know how too actually use it in the dial plan... The info on voip-info doesn't explain its usage very well... The dial plan example doesn't (to my mind anyway) specify an extention to call for conferenci...

Hi, i was successful in compiling app_conference and setting up an conference was quite easy. :-) Does anyone knows if it is possible to have an IVR accessable from inside the conference. So, if i dialed into an conference i want to be able to press '*' and then the actual discussion is muted for me and i and menu is read to me. Some...

I'm using Linphone. I tested with Asterisk and Speex only, I created a channel with echo and it worked. It seems to have problem when using app_conference. Jonathan 2006/1/31, Steve Kann <stevek@stevek.com>: > > jonathan blais wrote: > > > Hi, > > > > Does anyone ever used Speex with app_conference in Asterisk ? I'm > > having a hard time to figure why I always get this error "warning: > > Inval...

Hi, Does anyone ever used Speex with app_conference in Asterisk ? I'm having a hard time to figure why I always get this error "warning: Invalid mode encountered: corrupted stream?". Jonathan Blais -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/2006...

Has anybody compiled app_conference as of late? I've already asked on the app_conference devel list but as I'm rather in a hurry my thinking is somebody here has both run into and found a way to get this compiled and running. Using stable asterisk and the most recent app_conference from it's cvs on sourceforge.. ----...

...ified in the RTP draft. Jean-Marc Le mardi 31 janvier 2006 ? 10:43 -0500, Steve Kann a ?crit : > jonathan blais wrote: > > I'm using Linphone. I tested with Asterisk and Speex only, I created > > a channel with echo and it worked. It seems to have problem when > > using app_conference. > > If you just use app_echo, then asterisk won't be trying to decode your > frames; it will just be sending them back to you. Therefore, if your > client is using an incompatible packing of the frames, it won't > matter, as long as your client can also understand it. >...

jonathan blais wrote: > I'm using Linphone. I tested with Asterisk and Speex only, I created a > channel with echo and it worked. It seems to have problem when using > app_conference. If you just use app_echo, then asterisk won't be trying to decode your frames; it will just be sending them back to you. Therefore, if your client is using an incompatible packing of the frames, it won't matter, as long as your client can also understand it. In order to determine if...

...di 31 janvier 2006 ? 10:43 -0500, Steve Kann a ?crit : > > >>jonathan blais wrote: >> >> >>>I'm using Linphone. I tested with Asterisk and Speex only, I created >>>a channel with echo and it worked. It seems to have problem when >>>using app_conference. >>> >>> >>If you just use app_echo, then asterisk won't be trying to decode your >>frames; it will just be sending them back to you. Therefore, if your >>client is using an incompatible packing of the frames, it won't >>matter, as long as yo...

...pport video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video support to app_conference (also in iaxclient's CVS repository), and generally improve the iaxclient audio and codebase." So my guess from this is that there is currently no support for video in app_meetme, but that in the (hopefully not too distant) future, app_conference will be the replacement for app_meetme and...

Dear, I am using app_conference, 2.0.1, with asterisk 1.4. only a problem, if one of callers, disconnects the line, all of callers will be disconnected. and conference room will be removed. where is the problem ? best Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.di...

...onferencing. I haven't bought the machine yet so I would be interested what people used for hardware as well. >From what I have seen in the archives Jeremy scaled to 185 people with a Dual Xeon, but I don't know how it was configured or how much headroom was left on it. I am aware of app_conference but I would like to do this with meetme. Have a great day! Geoff -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 12/28/2004

Jean-Marc Valin wrote: >If you set the denoiser to "on" and the VAD to "off", what difference >does it make in CPU time? > <p>Same program, running on Athlon XP 1700+: Test 1, using VAD, but AGC, denoise off: tevek@canarsie:~/work/hms/app_conference $ time ./vad_test /tmp/demo-instruct.sw 5 reading from /tmp/demo-instruct.sw, repeating 5 times read 537760 samples beginning pass beginning pass beginning pass beginning pass beginning pass done. real 0m4.970s us...

...h, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference http://sourceforge.net/projects/appconference/ ?? THX Bye Alessandro R. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081211/15f83b48/attachment.htm

I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, s...

...presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has been changed." - http://www.areski.net/asterisk-meetme/about.php?s=0 Another variant is app_conference, that according to the author, can handle more simultaneous users: "If people are looking for a higher-capacity conferencing application, take a look at app_conference, in the iaxclient (on sourceforge) CVS. I haven't really benchmarked meetme, but I _think_ that app_conference...