similar to: recommendations for RJ-11 surge supressors?

I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through).... Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd

I think the time for me to get involved in NUT documentation has come again. Late last week I had to buy a UPS under time pressure. The Eaton unit that thus project gifted me with in 2006(?) died during a severe thunderstorm watch, so it was off to MicroCenter to get a replacement pronto. I wound up buying an APC BN700MC. It was what they had in the performance range I needed. The removable

I am sorry if this is an FAQ, but there doesn't seem to be searchable archives, and the supported manufacturer's list doesn't include "surge master". Google was also not much help. I have just put a Linux box and the customer has a "surge max " UPS. Anyone any ideas what protocol it uses? There is a Green serial cable. The manufacturers web site is here:

I am located in the UK, and I am using Sipura spa-2000 adapters to connect analog phones to a voip network. The network connects to the PSTN as well via the Sipura spa-3000 adapter. I would like to provide surge protection for the spa-2000 and the spa-3000 adapters. 1. For spa-2000, fxs port: What is the maximum tip-to-ring voltage before damage to the the adapter occurs? 2. For spa-2000,

I realize that the preprocessor echo suppression is not designed specifically to counter the effects of data loss. On the other hand, as much as I might like to, I do not have the option to "fix that problem for real." The sad truth is that some end users have systems that may drop samples, or do other unfathomable things, no matter what I do. I can not rewrite their drivers, firmware,

As previously noted, I need to replace a UPS and want to do my product research in public so others can benefit. The information I elicit on this thread will be merged into the UPS-HOWTO. I went to my local computer-equiuipment big-box store, a place called MicroCenter in Ardmore, Pennsylvania. I looked at the range of "UPS" devices they sell and was *completely* confused. Bear in

Believe me; I've "played with" priorities and buffering. Did you just say you have no idea how the Speex residual echo suppressor works? If that is the case, can you tell me where I could get some information about it? -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Friday, June 29, 2007 7:38 PM To: Coffey, Michael Cc:

Millions of ordinary people have systems that do not meet your stringent specifications. It does not matter what OS you or I use. It's the customers for whom this work is done. -----Original Message----- From: Jean-Marc Valin [mailto:Jean-Marc.Valin@USherbrooke.ca] Sent: Monday, July 02, 2007 6:35 PM To: Coffey, Michael Cc: speex-dev@xiph.org Subject: RE: [Speex-dev] Backup Echo Suppression

Just wondering if there's any significance to the jacks on the TDM cards. They appear to be RJ-45 instead of RJ-11, and I wasn't quite sure if that's something that makes a difference from the user perspective. . . Thx. B.

Hi everyone! Does anyone know how to install the rj package in Windows (7)? I am trying to set up eclipse, but I got [INFO] The R package 'rj' is not available, R-StatET tools cannot be initialized. Thank you in advance!

how to simulate a RJ 45 port to act as serial port any option in Centos 5.1. Please help me out on this. Regards, Gopinath M Signal Networks Pvt. Ltd. Smile... it increases your face value! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20080418/bcacfdf9/attachment-0001.html>

Jean-Marc, I think I can speak for most everyone here when I say we appreciate *very much* your code beyond the codec that helps people who use Speex in real applications. How far you want to take that code is up to you, of course. (And contributors, to a degree.) I don't think any reasonable person is expecting your code to solve everything. But when people are running up against

We're having an issue with connecting a Cisco ITS installation to * such that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or to any of the interfaces behind *. On the Cisco Side: dial-peer voice 8 voip destination-pattern 9999$ session protocol sipv2 session target ipv4:172.16.1.249 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad We have also

What is the current status of the residual echo suppressor in the preprocessor? I ask this because I have occasional problems with the AEC, in which it fails to cancel echo. I think the problems have to do with samples being dropped by the audio IO system, which is effectively impossible to detect and counteract reliably. A colleague recommended using an echo suppression technique (which he

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

I have a mixture network with WinXP machines and IRIX (UNIX of Silicon Graphics) machines. In the machine with IRIX (O2) there was an old version of samba. It worked well. But recently I've downloaded. I installed this new version, and I spoiled the system. Now I don't know if I have to reconfigure the smb.conf only or if I have to install more pakages or if I have to reconfigure several

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

All, I have a fairly old Compaq R3000h which is overdue for new batteries. Before I go and splash out a few hundred francs, I'd just need a quick sanity check - when I disconnect the power (UPS under load), it cuts out pretty much right away, so the batteries are flat (even though the charge indicator says full). When I reconnect the power, the surge triggers a 16A mains fuse. This sounds