similar to: app_conference, CVS HEAD, SIP and Xen

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

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. Something like the ${MEETME_AGI_BACKGROUND} in MeetMe.

Just curious, how does Asterisk pack Speex frames in a packet. AFAIK, Linphone just sends raw packets, as specified 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 >

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/20060131/386141a8/attachment.htm

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support 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

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:

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

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)?

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

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

Jean-Marc Valin wrote: >Just curious, how does Asterisk pack Speex frames in a packet. AFAIK, >Linphone just sends raw packets, as specified in the RTP draft. > > Asterisk expects speex frames to have a terminator. The phone I was referring to was the X-Ten/X-Lite phones, which seemed to be adding something _before_ the speex data to indicate the length of the frames.

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

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.. -------------- next part -------------- A non-text attachment was

I'm running into a bit of a problem setting up conference calls. The box I rent at a colo doesn't seem to have USB hardware.... When I try to load usb-uhci I receive a "device does not exist" error. Which means I can't load ztdummy.... The system has a rtc clock module, so zaprtc won't work... (which I'm scared to unload rtc because I don't have physical access

Hi, I'm still strugling with getting an easy to use conference system implemented. I did have app_conference running, but today I upgraded asterisk to 1.0.9 and it stopped working. I've tried following the instructions for compiling app_conference on 1.0.7 but it didn't work. So I went back to ztdummy (I've not had any luck getting this to compile on FC2). Anyhoo, I've

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:

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[sorry for the loss of proper attributions, this is from two messages]: [Me] >This is something I've encountered in trying to make a particular > asterisk application handle properly IAX2 frames which contain either > 20ms of 40ms of speex data. For a CBR case, where the bitrate is > known, this is fairly easy to do, especially if the frames _do_ always > end on byte

Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, 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