similar to: Compiling VD_app_conference for x86_64

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

On Fri, Apr 02, 2004 at 12:33:13AM -0500, Jean-Marc Valin wrote: > Do you have any sample code for that? Also, how do you tell autoconf to > append '-msse' without running into problems when CFLAGS is not set (and > usually defaults to -g -O2, but not always). Example patch attached. It only tries if the use passes --enable-sse; testing by target arch as Aron suggested is

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

On Mon, 2005-02-28 at 19:42 -0500, Jean-Marc Valin wrote: > > jean-marc: i think we can remove spx_sig_t *orig. > > but am not sure about exc2Buf. is it for extension? > > orig is already removed in SVN (which you should probably use). As for > exc2, it can be removed, but I'm not sure if you can just use exc > instead (maybe yes). > when removing "spx_sig_t

Hi All, Clang lacks support for the -fsingle-precision-constant flag. Are there specific reasons for this or is it just waiting to be implemented? This flag is especially important in the embedded world. From http://processors.wiki.ti.com/index.php/Floating_Point_Optimization#float_vs._double_vs._long_double : *Once all of your data is defined as float, there are still cases where you may

Alfred E. Heggestad wrote: >On Mon, 2005-02-28 at 19:42 -0500, Jean-Marc Valin wrote: > > >>>jean-marc: i think we can remove spx_sig_t *orig. >>>but am not sure about exc2Buf. is it for extension? >>> >>> >>orig is already removed in SVN (which you should probably use). As for >>exc2, it can be removed, but I'm not sure if you

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

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)?

I have Asterisk running 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

Thanks for the response. If we add the support would you accept the patch? Seems like a pretty straightforward flag since it maps directly to NumericLiteralParser::NumericLiteralParser within LiteralSupport.cpp. I understand the maintenance concern with flags that affect multiple points in code though. Still trying to get the bottom of why we're crashing with double floating point literal.

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

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

When compiling Speex 1.1.4 with GCC 3.3.2, the option -msse must be added to the CFLAGS in libspeex/Makefile. GCC 3.1.1 added a new option "-msse" (see http://gcc.gnu.org/gcc-3.1/changes.html , specifically under "New Targets and Target Specific Improvements") to enable SSE instructions within the compiler's output (for appropriate architectures). Compiling speex on

Hi everyone. I am looking at providing a conference for up to 300 people and was wondering if anyone has scaled meetme to 300 people. Here are some points: 1) I am using an IAX2 gateway hosted on a VOIP service provider. 2) The machine is hosted at the providers site so one has to assume that bandwidth is not going to be an issue. 3) Everything is coming in as ULAW so we won't need to

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

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, so that if I start recording

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