similar to: Static/Breaking up after I upgraded Asterisk aswell as a crash - Can't trace bug

If your remotes are not reporting any trouble. This may be farfetched but power may be to blame. I have had the ciscos 'freak out' with unstable power. It looks like the load on the power cubes cannot keep the caps loaded to deal with fluctuations. Or you may have a ground loop somewhere. Are the phones plugged into UPSs? I had flaky 7960 work fine after pluged into a Cisco POE switch.

We use Cisco 7960's with the P0S3-7-3-00 firmware, which was the latest as of a few months ago. I've so far found CVS-v1-0-10/26/04-07:28:01 to be the best version of Asterisk I've found. I've upgraded regularly in the past, like every other week. I upgraded to this version and also encountered no issues. About a month later I tried upgrading, to some version in November, and

I've compiled Asterisk on Redhat 9 and Fedora Core 1 in the past, generally without any problems. Especially w/ the stock kernel, which I generally loathe. When I tried to upgrade my Redhat 9 Server to the 2.4.27 kernel, doing a manual/clean compile, I had massive quality issues. I was forced to go back down to a stock 2.4.24 kernel. Never figured out why. Now, I've installed Gentoo

I'm trying to set-up Asterisk server and I would like to buy 2 SIP phones. Has anybody tried Cisco 7960G? Or 7940? What audio compressions can I use with this phone and Asterisk? Reason why I'm asking is because Cisco supports G.711 and G.729a audio compression (probobaly some tohers but they are not listed on data sheet) and on Asterisk features i found that it supports G.729 but need

Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't

-------------- next part -------------- An embedded message was scrubbed... From: Deon Rodden <drodden@webunited.net> Subject: Re: [Asterisk-Users] Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp Date: Thu, 30 Sep 2004 09:05:39 -0400 Size: 5509 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/289c69cc/dsp.eml

I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM. 15KRPM Drive. Using the default configs and added one Soft Sip phone. While listening to the demo the quality isnt very good. It's kind of crackly and skips a bit. Should the sound be better or is that just what you get using IP phones/Asterisk? (I ran the X-Lite phone). Nihal -----BEGIN PGP PUBLIC KEY

Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back! An overview: We have about 12 Linksys SPA941 SIP phones connected on a private switched network to our asterisk box which is a

Hi We have static noise problem on our asterisk server. latest stable release. The card is a new TDM04B We have it installed on the following hardware Motherboard Intel SE7520BD2SCSI 2x POWER SUPPLY 730W INTEL I will not mention the other hardware because we have desactivated/changed all the other items The only 2 items that we have not changed is the mobo and the power supply. At first it

Hi All, I'm currently running Asterisk with SIP phones and an ISDN card using chan_capi. I've just started to use IAX (GSM codec)over the Internet and the sound is adequate. However, there is an occasional 'glitch' in the audio resulting in lost sound or distortion. Is the distortion because I'm using zaprtc for timing instead of a zaptel card, or is more likely to be due to

Hi, There is any experience using Asterisk with VMWare? I think about installing a virtual linux box over VMWare and then Asterisk over it. Thanks, Dan

Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

Does anyone have any input into integrating asterisk with a alcatel 4400 PBX. Acording to what i've found is that Alcatel uses R2 for E1 -- regards Vikram (http://www.vicramresearch.com)

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Hullo. Having found http://bugs.digium.com/bug_view_page.php?bug_id=0000875 I grabbed the original 0.0.1 and Dan's patch, and whilst it didn't apply, I was able to patch the zaprtc.c manually - the Makefile has changed a lot, and I wasn't able to understand the changes. (this is all on a machine that's never had any * on it before, and has a 2.6.5 kernel with a matching

I finally got 2.4 recompiled with RTC as a module: Module Size Used by Not tainted autofs 13684 0 (autoclean) (unused) acenic 241092 0 (unused) iptable_filter 2412 0 (autoclean) (unused) ip_tables 15864 1 [iptable_filter] e100 62340 1 rtc 9084 0 (autoclean) Here is my

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

Hi all, I am working on building a new VoIP PBX. Looking at the current market for phones it seems my best "enterprise" options are the Cisco and Polycom phones. I have some experiance with the Cisco 7940G, but the process of flashing the phone with the SIP firmware left a bad taste in my mouth (not to mention the added expense for the phone). What is the general consensis about

Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: ***