similar to: app_meetme build option is XXX'ed out

> Ok, I don't expect everything would just work. But please run at least > SingleSource and MultiSource and let us know what breaks. Once it's > proven to pass a good chunk of the tests it can be checked in but > turned off. This will allow us to test it as llcbeta. > I just ran the test suite in llvm-test with a vanilla copy of the svn tree on darwin-x86. To perform the

Interesting. Are you using the included cbend.php script to terminate conferences? I occasionally get questions about using WMM with Confbridge, and to date I have not had an answer . If you can provide details, even vague ones, about how you did it, I can update the WMM package. Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at

Hi, I`m having a problem I cannot explain. When dialing 555-555-5555 (for example), I get a ringing sound until the person answers. When I have my Polycom forwarded to 555-555-5555, I do not get the ringing, but it dials fine and eventually when the person answers everything works fine. Where could be the difference? Both are using the same context to dial out. Mike --------------

On Sep 11, 2007, at 11:21 AM, Arnold Schwaighofer wrote: > Begin forwarded message: > >> From: Evan Cheng <evan.cheng at apple.com> >> Date: 11 September 2007 19:26:39 GMT+02:00 >> To: LLVM Developers Mailing List <llvmdev at cs.uiuc.edu> >> Subject: Re: [LLVMdev] RFC: Tail call optimization X86 >> Reply-To: LLVM Developers Mailing List <llvmdev

> > >> Currently, the first is called the "LLVM test suite" (which is the >> subject of >> the testing documentation). The second is called the DejaGNU tests, >> code >> fragment tests or llvm/test. The last one is the most confusing, >> it's called >> the whole program tests, test-suite (named after the svn directory >> it is in,

Begin forwarded message: > From: Evan Cheng <evan.cheng at apple.com> > Date: 11 September 2007 19:26:39 GMT+02:00 > To: LLVM Developers Mailing List <llvmdev at cs.uiuc.edu> > Subject: Re: [LLVMdev] RFC: Tail call optimization X86 > Reply-To: LLVM Developers Mailing List <llvmdev at cs.uiuc.edu> > > Hi Arnold, > > Thanks for the patch. Some questions

Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104 at sipphones:1] MeetMe("SIP/101-00000813", "104") in new stack == Parsing

Hi, I have asterisk installed in the xen virtual server. I installed zaptel 1.4.2.1 and patched it to have ztxen module. I loaded ztxen module but when I try to invoke or call to my meetme application I get the following warning and negative result of connecting to conference: [Feb 5 17:46:13] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:13] --

Hello, I've created a patch to correct error responses for the MeetMeList manager action. Currently MeetMeList produces an error if no conferences are active, success if any conferences are open. Requesting a conference that is not active while other conferences are active does not produce an error. https://issues.asterisk.org/view.php?id=18141 With the patch

Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian

asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that " WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension " . I found that there havn't "app_meetme.so" in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no

Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. >You have to compile and install Zaptel first, for asterisk to build meetme. > >-- >-- >Steven > >http://www.glimasoutheast.org > > > >"??" <lijun820311@163.com> wrote in message news:45C1B35E.0037E8.32263@m5-81.163.com... >> asterisk-users@lists.digium.com

Bill Gibbs,hello Thank you so much. According to this method , I get the "app_meetme.so" . ======= 2007-02-01 22:49:43 ????????======= >Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. > >-----Original Message----- >From:

Hi there, One of our Asterisk 1.2 machine is experiencing problems with MeetMe. Whenever meetme runs, the console is flooded with warning messages: The messages started as "No such file or directory" and becomes "Resource temporarily unavailable". I couldn't figure out what file MeetMe might be looking for, could anyone help? May 4 08:57:38 WARNING[19032]:

Is there an easy way to have app_meetme create the recording in a temp location and move it once the conference is over? or should I just have a perl script run every minute to check for no users in the conference room and then move it? Asterisk 11 Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I can't seem to use MeetMe app in asterisk versions beyond 1.8. Its source file app_meetme.c is present in the apps dir. Also, I can find app_meetme change-logs on the asterisk website. However, the dialplan doesn't have this cmd. I have checked menuselect but it says it has been replaced by app_confbridge. Also, If that *is* the case, does ConfBridge (the newer version of meetme)

http://bugzilla.mindrot.org/show_bug.cgi?id=647 Summary: Setting "UsePAM no" in sshd_config gives error if not config'ed w/ --with-pam Product: Portable OpenSSH Version: -current Platform: Sparc OS/Version: SunOS Status: NEW Severity: minor Priority: P4 Component: sshd

I originally built my Asterisk server without installing the Zaptel package as it was going to be a purely SIP based system. However when I went to setup conferencing using meetme I found out that app_meetme is dependant on the ztdummy for timing. I have now installed the zaptel package and I believe the ztdummy module is loading ok [root@astro asterisk-1.4.0-beta2]# lsmod Module

Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject:

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17