similar to: Debugging a SIP / AudioCodes Problem

I have a Gamatronic Smart Compact 750. I have it connected to my linux machine via serial cable and am running nut 2.0.3. There is no interruption in power, but the ups is reporting OB (on battery). Is there a configuration option I am missing? The relevant ups.conf entry is: [Gamatronic] port = /dev/ttyS1 desc = "Home Gamatronic UPS" driver = genericups

I was told by someone who used to have a Gamatronic to use the Fenton driver. However, when I use the Fenton driver I receive the following error: Unknown ups - please report this ID string: #GAMATRONICS SC06GM0G V8.0 Detected Unknown C06GM0G on /dev/ttyS1 The person who pointed me at this driver mentioned that Gamatronic changed their banner and a patch or code change is needed. I

I''m writing a method_missing handler to "automagically" create assertions from predicate functions. So, for example, one use might be: assert_not_include ''joes@shmoe.com'', response.to -or- assert_include ''info@mydomain.com'', response.from The code is working properly, but when I trigger an assert, the backtrace shows something like:

I have a spec: specify "a post with no user id should add a record" do post :edit_or_create, :user => {:login => ''joeschmoe'', :email => ''joe at shmoe.com'', :email_confirmation => ''joe at shmoe.com'', :full_name => ''Joe Schmoe''} assigns[:user].should_not_be_nil

Il 20/08/2013 13:13, Naor Shlomo ha scritto: > Hi Paolo and thanks for your help. > > I upgraded the following (compiled from source) > qemu : 1.5.2 stable > libvirt : 1.1.1 > > but for some reason when I run the version command inside virsh: > > Compiled against library: libvirt 1.1.1 > Using library: libvirt 1.1.1 > Using API: QEMU 1.1.1 > Running

Hello, I'm using a 64bit Linux with 16GB of RAM. I'd like to limit the memory that the R process can use so I'm trying to use --max-vsize switch. However, it is seems that I can't enforce a limit above 2GB. shlomo at hippo:~$ uname -a Linux hippo 2.6.24-16-generic #1 SMP Thu Apr 10 12:47:45 UTC 2008 x86_64 GNU/Linux This WORKS: -------------------- shlomo at hippo:~$ R

Hi. I recently upgraded to R version 2.4.0, and I have found that the "debug" package no longer works. In particular, when I try to debug a function, I get the following error message: Error in attr(value, "row.names") <- rlabs : row names must be 'character' or 'integer', not 'double' I have, of course, taken all the necessary preceding

Hello Users: I'm using R version 2.0.1, and having problems with RODBC. Everything works fine when I use Rgui, but when I try to use Rterm and issue the commands library(RODBC) con <- odbcConnect("MySQL", "test") I get the following error: Error in sqlQuery(con, str) : first argument is not an open RODBC channel In addition: Warning messages:

hi we are trying to decode speech that was encoded on a ds'pic with a NB 8 Kb/sec. we wrote a simple program using the API speex lib 1.2 beta 3: #include "stdafx.h" #include <iostream> #include <stdio.h> #include "speex.h" #include "speex_bits.h" using namespace std; char hex2ascii(char toconv); #define FRAME_SIZE 160 #define nbBytes 20

hi we are trying to decode speech that was encoded on a ds'pic with a NB 8 Kb/sec. we wrote a simple program using the API speex lib 1.2 beta 3: #include "stdafx.h" #include <iostream> #include <stdio.h> #include "speex.h" #include "speex_bits.h" using namespace std; char hex2ascii(char toconv); #define FRAME_SIZE 160 #define nbBytes 20 int main(int

Many useful Netscape plugins don't come in Linux versions. Is it possible to make my Linux copy of Netscape use Windows-compiled plugins? If so, how?

I have an Audiocodes MP-118FXO in production. When an outbound call is made and the remote party hangs up, the Audiocodes hangs up the call immediately. But if an incoming call is received and the remote party hangs up, the Audiocodes does not hang up immediately. I have tinkered with Current Disconnect and Polarity Reversal settings, to no avail. Anyone experienced this issue with Audiocodes or

I have an Audiocodes MP-114 and modem dial-out can not establish connection. Snooping with Wireshark I get a lot of entries like this: No. Time Source Destination Protocol Info 2934 27.221559 10.0.0.110 10.0.0.109 RTP PT=ITU-T G.711 PCMU, SSRC=0x7D08611A, Seq=7662, Time=1805957855 Frame 2934: 214 bytes on wire (1712 bits),

Hi, This is my first experience with a mediant 1000 and an Asterisk Trixbox. the mediant has 12 FXOs and 12 FXSs, and I want to use it them all. I will have extensions connected to the FXS ports, and lines to the FXO. Can anyone guide me, please? regards, -- Guillermo Garron "Linux IS user friendly... It's just selective about who its friends are." (Using Ubuntu, Debian,

I have an MP114 2fxs,2fxo which I would like to use with Trixbox, does anyone have a setup file they can share to help me work this out. Instructions or a link I can follow - thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090726/643e5b96/attachment.htm

I'm trying to define distinctive rings for lines in this gateway but don't works. Nothing happen when sending a call...the phone doesn't ring.... The same configuration works fine for PAP2NA devices. Adriano Almeida Flickr agora em portugu?s. Voc? clica, todo mundo v?. Saiba mais. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sir, Here is the working Audiocodes MP-11X FXO configurations to work with Asterisk. ;************** ;** Ini File ** ;************** ;Board: MP-118 FXO ;Serial Number: 905371 ;Slot Number: 1 ;Software Version: 5.00A.024 ;DSP Software Version: 204IM => 209.13 ;Board IP Address: 192.168.0.195 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 192.168.0.1 ;Ram size: 32M Flash size:

Razza, I have a MP114 FXO/FXS that I have never got to work , even as an FXS, even though I have several other FXS's that work fine ie Linksys PAP2 etc.. would you put up your config? PDE

Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant Audiocodes 1000 where we have some fax machines connected to fxs ports, what we need is to make fax machines through mediant

Hello all, I'm working on a little project right now and have ran into a snag. Was hoping someone would be kind enough to give me a few pointers to help me get past the current issue... I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...) that I'm trying to get to play nice with Asterisk 1.4. I've got it to the point where the AudioCodes box picks up