similar to: asterisk knows best? softphones

Dear users, I am attempting to plot an ols() call from the Design package, by following the procedure explained by Harald Baayen in his 2008 book 'Analyzing linguistic data. A practical introduction to statistics using R', page 175-181. I've attached my data to this e-mail (I hope it's small enough that that's ok). First I paste all the commands I ran, followed by the

Not sure if it's alright to talk about this here??? compiled the STUN server from Vovida on RedHat 7.3. Looks simple to configure. It isn't starting...it tries to for a long time and then just craps out. Here is my config:/etc/sysconfig/stund #!/bin/echo Not to execute. # Path to stund STUND=/usr/sbin/stund # Set the required args for STUND STUNDPRIMARYHOSTNAME=208.x.x.x # The hostname

One for the gurus.. I have seen there has been a lot of discussion about using SER with Asterisk.. This to me seemed like an over kill becasue it would basically be doing most of what Asterisk is doing anyway unless you create some weird and wonderful config in SER.. Anyway, I decided to go and have a quick read through the SER docs and in the section about NAT they say that the best way to

No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I thought Asterisk was cool by itself, but Trixbox has made just about everything turnkey. Great stuff! So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox, which sits on our DMZ with a single public IP. I need the phones to work from random places behind NAT, as well as in the office. I'm using

Hi, I noticed some issues with how grandstream handles stun test. GS is running version 1.0.4.50. First I reset the NAT router. Then reboot GS, get results of "restricted cone". Immediately reboot GS, get results "full cone". I tried quite a few public and commercial stun servers. Also tried different model/version of linksys routers. I always got the same issue. Winstun on

Hi all, We've got a department that has 5 phones using a * 1.0.9 box. They need to have an extension that rings all 5 phones at the same time. Getting all of the phones to ring isn't a problem, but they are running into a problem with the phones ringing in their ears when they are already on a call. Example: Caller one calls the queue, all of the phones rings, and employee one

Guys. Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of ways to get around nat but I would like to hear some success stories about handling nat users with multiple voip phones behind nat. I have my asterisk box behind but ports are forwarded (5060 5004 10000-20000 for rtp and 4569 for iax2) but still.. I can quite figure out what ser and stund have to do on this

Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

I have an XP home (falcon) and an XP pro (tatooine) machine, both behind a rh9 box (hoth) acting as firewall. I've been trying to get samba set up on the linux box. I think I have it mostly working except that the XP boxes won't list hoth under "My Network Places - Microsoft Windows Network - MSHOME", and I cannot access hoth by name at all from the XP-pro box (by IP, I can

Yes, try http://1ezphone.com its a browser softphone. ----- Original Message ----- From: Zoa To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] SIP Softphones and Citrix ? Date: Fri, 01 Feb 2008 23:09:56 +0200 I'm working for zoiper.com and i'm willing to help out with ours when needed. Zoa d4rk f1br wrote: >

Hi all I'm looking for a softphone that works well under terminal services environment, we need to set up 24 to 32 phones for a call center, also, does any one knows if it will actually work fine under load?

Hello, A little Off-Topic but how can you easily switch microphone and headphone from one softphone to another on a Linux KDE platform ? I use Skype (for incoming calls mainly) and Twinkle (for outgoing and incoming calls). I could't find any practical way to quickly switch audio from one application to another. Ideally, I would like to "register" some kind of virtual audio

I am presently configuring a test Asterisk 11 server based on CentOS-6 and I need to employ a softphone for testing. The base repo has ekiga. The EPEL repo has twinkle. I lack the knowledge of whether other packages exist or might be better suited. Which of the two do you recommend? Or, alternatively, what other package might be a better choice? -- *** E-Mail is NOT a SECURE

Hi All, I'm currently using SJPhone as a sip client with Asterisk. It works perfectly with a USB Yap Phone but there is was slight problem! If I setup SJphone to use the USB device for audio the ringing tone is also over the USB device. If I'm away from my desk I can't hear the phone ring! Is there an option to use one audio device for calls but another for the ringing sound?

Hello, I'm new to Asterisk, and am trying to get the basic features under my belt until I move on to the more advanced ones. Currently I have two softphones registered with my * server on my network, and one of the phones can call the other just fine, but when I try to call from the other phone, I get a message on the phone saying that the call was terminated for reasons unknown.

Hi, I've been messing with getting SIP working for days now, with limited success. I've got Asterisk set up on a remote server with the echo test. Please try it out to verify I've got the server working right: sip:robot@nixon.butchwax.com Running FC1, ThinkPad T22, headset thru the soundcard. Asterisk is asterisk-1.0_RC1. No NAT. The phones I've tried so far are as

Has anyone tried Xten's new eyebeam SIP video softphone with Asterisk? http://www.xten.com/index.php?menu=products&smenu=eyebeam Jim James H. Thompson jht@lava.net

Hi I tried Xten, its very good, because it can stay in the taskbar (next to the clock) and start when windows starts, and is allways ready to receive calls. Maybe it s the best way to introduce VoIP to my company workers.... But theres a feature that s missing (or I couldnt find), there s no way to connect this softphone with the adress book. I think this feature is very important, because