similar to: Help with compilation of app_conference in x86_64

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

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 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

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

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

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.

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

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)?

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.

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

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 >

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:

Hello, I can?t compile "iaxclient", because one needs to compile the new version "wiax.dll". I tried to compile it under MinGW/Cygwin, but I had the messages like: cc -I. -Igsm/inc -Iportaudio/pa_common -Iportaudio/pablio -Iportmixer/px_common -Ilibspeex/include -g -O2 -DSPEEX_PREPROCESS=1 -DNEWJB -Ilibiax2/src -IAXC_IAX -DLIBIAX -DSPEEX_EC=1 -DWIN32 -DBUILDING_DLL -c

Hi I use RH 9 + asterisk v1.0 stable + 2 PCI fritz cards + chan_capi 3.5 and it works fine, Since my users want fax fonctionnality and customers know 1 of the msm as fax number I wanted to try the chan_capi-0.3.5 patch if I patch chan_capi and run make, I get an error message , as you can read below, orginal chan_capi compile works, when patched I get an error, no CID ? Any idea ? anybody

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

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

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