similar to: Re: why there havn't"app_meetme.so"fileaboutasterisk1.4.0?

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:

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

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

Hi, I have declared my sip users call-limit=2 and type=friend. When any user recieves a waiting call while already in a conversation, the peer call counter is set to 2.The problem is that, the counter is not reset to zero after hangup and becoz of this the user is not able to recieve any call anymore even if s/he has hungup. the asterisk cli displays the following error. [Mar 17 16:15:10]

Hi,All, I wang to use AGI in asterisk1.4. AGI file / myperl.agi #!/usr/bin/perl use strict; ...... print STDERR "7. Testing GET VARIABLE..."; print "GET VARIABLE EXTEN \"\"\n"; my $result = <STDIN>; &checkresult($result); ...... when the agi execute; asterisk conosle show that : AGI Rx << GET VARIABLE EXTEN "" AGI Tx >>

Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting

Hi, when I build zaptel-1.2 and asterisk-1.2 I can modprobe ztdummy and start asterisk to be able to use MeetMe(). When I build zaptel-1.4 and asterisk-1.4 I can modprobe ztdummy and start asterisk but I am not able to use MeetMe(). What do I miss? Stefan -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new

Same problem for me. In asterisk1.4.0 directory solved by; configure --with-zaptel=../zaptel-1.4.0 make make install -- Martti Tienhaara (martti@dash.ca) DASH Software Ltd.

HI,ALL, I have multiple PSTN lines registered as multiple SIP channels (e.g. SIP/line1, SIP/line2, SIP/line3, etc...), on the multiple gateways( I uses the SIPURA3000). I wants to arrange them into an ordered hunting group for outbound calling. I used http://www.voip-info.org/wiki/view/hunt-dial+macro for reference. My configure files like blew.

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

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)

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

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

Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I