similar to: Fwd: [Announcement] Asterisk-IL mailing list

> -----Original Message----- > From: Gilad Ben-Yossef [mailto:gilad@codefidence.com] > I'm prbably stupid, but wont this do what you want? > > > exten => 1,1,Goto(bye,s,1) No, because I wanted to match on "D", not "1". Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't

Redhat 9 is running 100% cpu usage. I had a couple boxes doing this. upgraded to Fedora and its ok.

BR C.

> -----Original Message----- > From: David Brodbeck [mailto:DavidB@mail.interclean.com] > Okay, the problem appears to be that I'm tone deaf. ;) > > I finally thought to turn on debugging on the channel. The > PBX is sending > "D", not "*". The programmer of the previous voice mail system (whose > configuration I was cribbing from) seems to have

Hello, I use Asterisk 1.0.2 on a RedHat Enterprise Server 3.0 (Kernel 2.4.21) and i experienced that the memory consumption of the asterisk-process started by the init.d-script raises continously. Now, after 3 hours of operation (on our testing-system we have 30 concurrent connections to another asterisk box using IAX2 and GSM codec) there is already 66MB allocated. I think this could be ok, but

At time to time somebody is trying "their luck" and send me most likely a junk fax on my voice line. During normal working hours is not a problem I just pickup the line and hangup the call but after-hours my voice mailbox is intercepting the call and recording those "beeps" (waisting my CPU cycles). Is there a way to block call / issue hangup command if the incoming call is a

Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100

I have 4 gig in my * box. I'm tuning for performance and I'd like to ask opinions: 1. asterisk -p == renice -20 ?? 2. I've turned off swap with no apparent ill effects. Can anyone commment on long term effects with moderate load (say, 30 SIP phones / 2-3K calls /day) 3. Can anyone comment on using ramdisk as swap and whether this is a good idea or bad idea? I'm using 2.6

By default asterisk (on my system) runs with NICE value "0" like most of the programs. Will the performance improve if I assign NICE value of let say -5 or -10 to asterisk? To my understanding it should as the priority will be higher than most other programs (it might even solve some echo problem). Am I right? How to start asterisk with priority let say -5? If I start the main process

from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx "restart now" >/dev/null 2>&1 or 10 7 * * * root /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1

Hi, I've switched the mailing list manager for the tinc lists from Majordomo to Listar today, for the users this change is fully transparent and for the list owners it just means bounce messages are handled by the system instead of by hand. The only minor change is that you may need to adjust your mail filters since the "Sender:" field now points to

Hi, I've switched the mailing list manager for the tinc lists from Majordomo to Listar today, for the users this change is fully transparent and for the list owners it just means bounce messages are handled by the system instead of by hand. The only minor change is that you may need to adjust your mail filters since the "Sender:" field now points to

On Sun, Jun 29, 2014 at 11:39 AM, Yossef Efraim <yossefe at mellanox.com> wrote: > Hi all ? > > Have any one successfully worked with pxelinux 5.10 and above ? > > Looks like ipxe/syslinux got broken at that point ?. Yossef, as I just mentioned about an hour ago, I've had mixed results. With an iPXE ISO, I've had no difficulties but hopefully about to try to test an

Hi all, Here's a brief history of my adventure: I purchased a CyberPower 685 AVR and decided to connect it through USB to my Gentoo linux machine running kernel 2.6.22. While the device was detected properly and added to /dev, it was added with the wrong permissions (gid was set to 'usb' instead of 'nut', as it was expected to be according to the rules under

Hi Jean-Marc, thanks for the response. First, I will clarify again that floating-point solves this - so isn't that a bug in fixed-point? Also, I understand that algorithmically the AEC won't cancel echo properly on a non-linear signal, but why completely distort the output? If the echo just won't get cancelled it would be acceptable, but in the current state it disables the ability to

(Viral: I'm forwarding your message to the FLAC mailing list) ----- Forwarded message from Viral <viral@debian.org> ----- Date: Fri, 3 Aug 2001 00:00:07 +0530 From: Viral <viral@debian.org> To: mdz@debian.org Subject: [rcw@debian.org: Re: Patches to abcde to include support for flac format.] Hi, I patched abcde so that it can encode flac files. The patch will not be included in

Hi Jan, I asked almost the exact same question as you about 6 months ago and someone provided some Gen4 results for me (But I can''t seem to find them in email), they were a fair bit better than Gen3. With IPOIB, you want connected mode and a large 32Kb+ MTU for max bandwidth, 1Gbyte/sec or more should be possible with Gen 4. Here are some of my original test results on my Opteron

Hi, The Speex AEC is simply not designed to deal with non-linear echo, as is the case when clipping of AGCs are involved. Make sure all your path is linear or forget about the Speex AEC. Jean-Marc On 11-01-03 11:31 AM, Omer Gilad wrote: > Hi, > I couldn't find a discussion that specifically addresses this, so here > it is. > > I'm using Speex AEC in my mobile VoIP

Do I have any other options besides RH 9.0 ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com <http://www.successfulhosting.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041128/ac580fa1/attachment.htm