similar to: call files

I am running asterisk 1.4.18 trying to use MeetMe and option b. I am getting permissions denied failed to execute conf-background.agi on the CLI lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi -> /home/silentm/bin/conf-background.agi my conf background is a symbolic link - then my permissions are : [root at devcentos5x64 src]# ls -l /home/silentm/bin/conf-background.agi

Is there a way to use kvm-amd on centos 5.1? I dont want to mess with XEN. I want the hardware virtualization that is on my AMD chip. I played with putting 2.6.24 and centos 5.1 on my AMD laptop, got kvm-amd and that works. However I want to put this on my desktop that is running 2.6.18-53.1.14.el5. Is there a way or do I just wave to use 2.6.24 or 25. Thanks, Jerry

I am reading comments on the Wiki for meetme http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe from 2004 about how and AGI does work with non zap channels. Is this still valid 3 years later and 1.4.4? How do I bring people into a meetme and play a message to all of them when they are on SIP channels? Jerry

How do I stop the "you have been kicked by this conference" message from speaking? I first had MeetMe(conf, l) and I get the kicked message. I tried Meetme(CONF, lq) and I still get he kicked message. and it still says it. Thanks, Jerry

Hiya, hope I don't bore anybody with this. There are certainly a lot of monitor-y things out there and they just didn't fit my need, so maybe this will fit someone's besides mine. http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one is a php script called pbxmonitor, and one is a flat file of extensions to extension name mappings of internal users. It

?? Hello.? We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).? We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.? If a person starts talking they will cut off others on the call.? Is this normal behavior?? Are there any options I can change to change this? ?? Thanks! James -------------- next

What does it take to get ztdummy to work correctly? I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 1.4.9.2 Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs. Problem is playback() does not work. So then I stop zaptel, asterisk runs and playback() now works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for that. I am

I am hearing a you have been kicked from the conference message in asterisk 1.2.23. I dont want to hear that. I am using 1qt for the meetme. How can I disable that message? THanks Jerry

Hi, is there a way to use g729 in meetme? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070929/74f6e5d9/attachment.htm

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

Is there a way in Asterisk (I guess there's, it's only I can't figure out how :-)) to: 1.- Generate a call to channel 1 (example, to PSTN v?a an E1 card, using Zap/g1) 2.- Generate a call to channel 2 (example, an internal SIP extension). 3.- Once both channel have answered, connect the call between them. This way, I can, for example, play audios in both channels before they are

I am using usb-audio for Console/Dsp with asterisk. it is crashing 1.4.21 and also svn. During the brief times its working the audio is choppy but understandable. I have used aplay and arecord at the same time on the same wave file and they work fine every time and I have done it MANY times. Asterisk failes after 1 or 2 times. Any ideas on something I can try? Jerry

I have used every asterisk 11.8.X version. Have not had an issue till 11.8.0 and 11.8.1 When I use ConfBridge I am attempting to put all participants in MUTE mode and just one talker... However, since 11.8.0 I am hearing feedback in the announcement like the channel is not really muted. I dropped back to 11.7.0 and I hear no feedback. Has something changed - or - am I not correctly setting up

hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305]

I am sending an email from my machine devcentos5x64. the transcript below (hangs for 60 seconds) at the line: MAIL From:<root at devcentos5x64.msgnet.com> SIZE=56 AUTH=root at devcentos5x64.msgnet.com The email succeeds - but I am trying to figure out the 60 second delay. Neither email server is busy. Nothing is waiting. the DNS on both machines point to the same nameserver. The DNS

hi folks, in my (unlucky) first test of ogg vs other encoders, i found a case where wma and mp3pro sound much better than ogg at 64k. can anyone suggest a setting that i haven't tried yet that can rival the wma and mp3pro samples at 64k? it's the "gravel effect" that is troublesome. the part in question is the first 15 seconds of this wave file:

To continue here with problems on this compaq v2000 laptop, I put kernel source on a USB disk and plugged it into the v2000. NOTHING IS recognized. I tried to manually mount the disk and nothing either... I thought USB was well established.... I thought trying to recompile the kernel for realtek support might get my networking going... I am stuck??? Jerry -------------- next part

Hi, I'm using Asterisk@home and am having trouble using the conference bridge that comes built in. We're using Polycom phones. When we transfer the first person into the conference room (e.g. 8101) , they get into the room fine. When we try to transfer a second person into the conference room, they get dropped as soon as we finish the transfer. This is using Polycom SoundPoint 301

hi all, running centos 4.7 i686. I seem to have an o pen r elay sendmail server. How do I close it? I have the STRAIGHT centos install sendmail.mc file. Only thing I changed was: dnl DAEMON_OPTIONS(`Port=smtp,Addr=127.0.0.1, Name=MTA')dnl so as to allow incoming email and not just localhost. however this seems to relay everyone. I looked at http://www.sendmail.org/tips/relaying but it

I am trying to hwclock to set the time. (hwclock -w) this is what I get on standard 5.2 x86_64. hwclock --debug hwclock from util-linux-2.13-pre7 hwclock: Open of /dev/rtc failed, errno=19: No such device. No usable clock interface found. Cannot access the Hardware Clock via any known method. [root at devcentos5x64 src]# ls -l /dev/rtc crw------- 1 root root 10, 135 Feb 6 13:32 /dev/rtc Any