similar to: Slightly broken audio on USB headset?

I'm trying to find a voip-suitable USB headset (I.E. headphones + microphone) which I can use with my laptop while I'm traveling and using Firefly or another softphone. I'm currently using a Logitech headset which works well (except the echo it generates toward the other caller when I turn up the gains too high), but it just doesn't carry well - in fact, I can't carry it in my

In article <CABr8-B6fcNgogynq66nNMkSLCHAtqA4OrL09XY6ABPRV4H+ZQw at mail.gmail.com>, Jerry Geis <jerry.geis at gmail.com> wrote: > Hi All, > > What is the correct way to provide a CentOS 7 - WMware image for ESX ? > > As an amateur to VMware - I thought - great I can get VMplayer and ESX > should be able to import my image... Wrong... I even went through the >

Hi, I'm using OH323, mostly with success, to interface Asterisk to a provider's switch (World Telecom INX). I have noticed a particular effect, and I wonder whether anyone else has seen the same? The effect is audio flutter (almost like the flutter one gets on MF or HF radio sometimes) which only happens intermittently. Audio coming into Asterisk is unaffected, as proved by using the

This question is directed towards those who are familiar with the inner workings of the Asterisk code. I'm quite at home hacking on the source code, and have become familiar with certain parts of Asterisk's operation. I'm looking for some advice on the most fruitful avenues to explore in order to achieve a particular application I need: either in the source code or in AGI (with which

I'm attempting to get chan_h323 working on Asterisk CVS-STABLE. I've compiled it ok using the Janus release of pwlib/openh323, by editing the makefile as per the comments. Call setup and cleardown seems to work fine, but no audio is being passed in either direction. Doing an "h.323 trace 9", I noticed the following sequence at the end of the call setup: h323.cxx(1685)

I have a customer with an Asterisk 1.4 system (r144238 - between 1.4.22-rc5 and 1.4.22 released). It uses SIP to connect to the PSTN via a provider who is on the same LAN as the box (it is co-located at the provider). They also have about 20 SIP phones as extensions that connect to the box over the internet. "sip show peers" indicates that most phones have a latency of 90ms-100ms. The

Hi, I'm trying to diagnose a particularly elusive problem, and am wondering if anyone else here has seen anything similar and can offer any ideas. I have a conference bridge running Asterisk 1.2.32 (with slight mods), in a colo talking via a LAN to an ITSP using SIP/RTP. It is dedicated to a single customer. On several occasions over the last few months, the customer has reported instances

On 06/05/2015 03:48 PM, John R Pierce wrote: > On 6/5/2015 1:44 PM, g wrote: >> for what it would take to do so, you are better off getting buying >> better quality at a little more cost. > > I've had pretty good luck with the basic models of Plantronics > headsets. They tend to be well made. The Logitech stuff I've > bought has often broken within a

Hi, Can anyone recommend a USB phone that can be used under Linux, either interfacing directly with Asterisk in some way, or using a soft phone program on Linux that doesn't need screen interaction (only using the phone's keypad)? The idea is to be able to plug it into the USB port of an Asterisk box in a rack, where screen, kbd and mouse may not be available. Thanks in advance! Tony --

Hello Everybody I got some little issues with the sound. Logged in as my normal user - I've the testsound (winecfg) on the internal device. I can't change it. If I log in as root I've the testsound on my external USB speakers (the way I would prefer it works). I can't Imagine me - why there is a different between root and my normal user. Please find below some details. Code:

Hi, we are looking to roll out PBX IN A Flash at our office. The first group will be using Soft Phones (X-Lite appears to be the best and works in Windows, Apple & Linux). There are many types of USB Headsets to choose from and a fairly broad price range. Is there any USB headsets people would recommend? I'm specifically interested in acceptable audio (speaker and microphone) quality

In article <mfbt6f$9rt$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that > is behind a network device to which I don't have ready access, which is > performing NAT with possibly some kind of SIP ALG, and an Asterisk 11 > system on a public IP. > > My question is

In article <n009u2$85v$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > Apologies, this is slightly off-topic being to do with an EPEL package, > although it's running on CentOS6, so I thought others here might have come > across this issue. > > I have five CentOS 6 systems running fail2ban from EPEL, and this > package was updated

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

On 6/5/2015 1:44 PM, g wrote: > for what it would take to do so, you are better off getting buying > better quality at a little more cost. I've had pretty good luck with the basic models of Plantronics headsets. They tend to be well made. The Logitech stuff I've bought has often broken within a year. -- john r pierce, recycling bits in santa cruz

RJ11 Plantronics 1 3 2 4 3 1 4 2 RJ11 Pin 1 is on the left when looking at the contact points. Plantronics Pin 1 is on the left when looking at the contacts (through the plastic sheild) My multimeter battery is low, so YMMV, but: Pin's 1,4 are connected with ~160 ohms Pin's 2,3 are connected with ~1400 ohms On my Plantronics head set. Chad -----Original Message----- From:

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

In article <ml1jnh$afr$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > When I started using CentOS 6 instead of CentOS 5, I discovered that > "less" no longer understood \< and \>, which I had been used to using > since almost forever. > > Eventually research revealed that in the Fedora version on which > RHEL 6 was

Thanks for the suggestion Tony, I installed each codec for MoH, core sounds, and extra sound packages. Unfortunately the tests produce the same results. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 ( continuously for a while followed by a [Sep 1 20:36:46] WARNING[7761][C-0000770d]:

worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -----Original Message----- From: tony@softins.clara.co.uk [mailto:tony@softins.clara.co.uk] Sent: Tuesday, April 20, 2004 12:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In