similar to: kphone and *

Hello, I'm having a problem getting two (working) NICs enslaved to a bridge to both work in the bridge. The setup is just for testing and looks like this: (NETWORK)-------|bridge|---------|test PC| I'm building a bridge from an old Dell Optiplex GX1 desktop. Using kernel-2.6.9-gentoo-r4, with bridging and ebtables compiled into the kernel; the two PCI NICs are Intel, and the e100

Hello, I know that my Kphone question may be a bit off topic, but I have been busy with this again and again for about one month now, sent three mails to kphone@wirlab.net (the contact address mentioned on http://www.wirlab.net/kphone/index.html), asked for a solution in a german ip phone forum and tryed many things by myself. I try to compile KPhone 4.0.3 (tryed CVS Version as well) but

hello, I am running asterisk on one linux PC and want to talk through this server using Kphone installed on 2 different PC's. These are the extra lines added to sip.conf and extensions.conf respectively. sip.conf [jitha] type=friend host=dynamic secret=jitha context=sip dtmfmode=inband [sudhananda] type=friend host=dynamic secret=sudhananda context=sip extensions.conf [sip]

Hi, i just installed latest asterisk on fedora rc2 and on the same machine i installed a sip soft phone called kphone. Kphone complains about /dev/dsp being used and can't place/answer calls (/dev/dsp is obviously used by asterisk) . how can "share" my sound card with these two programs? or can i disable the sound card in asterisk so i can use kphone to place/answer calls? BTW kphone

Dear Asterisk experts, I've just downloaded Asterisk CVS version (since I'd like to manage its configuration from RealTime). Next, I have kphone on the same Linux host, and VmWare virtual machine with Windows and X-Lite IP phone inside. I successfully tested the demo's with X-Lite, but failed to hear something with kphone (v4.1.1). There were NO problem with this kphone and stable

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK

I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>

Hi, I am new to Asterisk and SIP. I have just installed ser as sip server and asterisk ser is in 192.168.6.244, without authentication kphone as sip client in 192.168.6.254 asterisk is in 192.168.6.100 and did not install hardward on my pc in sip.conf , i add following lines ... register:jimmy@192.168.6.244/1000 ... [192.168.6.244] type=friends host=192.168.6.100 fromuser=jimmy

here at wkdu.org our minds are blown trying to figure out what our problem is, we are using a dell optiplex gx1 with a soundblaster live card. we are getting a nasty hum on the line in. the thing is we plug a cd player directly into the soundcard the hum goes away and when we plug the line in into anything else there is no hum, so the question i ask where is the hum coming from? we are

On Wed, 24 Apr 2002, Alex Walker wrote: > On Tue, Apr 23, 2002 at 01:16:34AM -0400, jim wrote: > > here at wkdu.org our minds are blown trying to figure out what our > > problem is, we are using a dell optiplex gx1 with a soundblaster live > > card. we are getting a nasty hum on the line in. the thing is we plug > > a cd player directly into the soundcard the hum goes

hello all, i have already configure sip.conf and dialplan. i done the follow me script. first problem: i want to call(with kphone) someone at my extension, i must dial the extension number. i can't dial their username. 20531603@192.168.8.125 (work) mustafa@192.168.8.125 (call fail) is it possible to do that?? second problem: if i want to call another number (not my

hi all when starting kphone, it tries to register with asterisk but fails after a while. The SIP entry in * for this user is below. This is identical to the other SIP entries. The other SIP clients are MSN messenger plus one snom. these work fine. See SIP debug output attached as 'screen-exchange' thanks roy [roy] type=friend ;insecure=yes username=roy ;secret=password host=dynamic

Hey, I have a problem booting certain machines with pxelinux. I have pxelinux.0 in the root of my tftp server and dhcp pointing to the right location: # dhcpd.conf authoritative; option domain-name "example.net"; ddns-update-style none; default-lease-time 3600; max-lease-time 86400; subnet 10.0.0.0 netmask 255.255.255.0 { range 10.0.0.3 10.0.0.20; server-name

I'm trying to get two linux machines with kphone-3.11 two communicate with each other over asterisk. I have them configured correctly on asterisk to use sip channels, but when I call from one phone to the other I don't any voice communication between the phones. According to the phones I'm connected, but according to asterisk, I get the following message: -- Executing

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

Got no responses to this, but the list seemed to be down for a while, so here it is again. Sorry for the extra bandwidth! John 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 at nixon.butchwax.com

Hi all, i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) and this is what my "sip show users" return: moloch*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No

On Tue, Apr 23, 2002 at 01:16:34AM -0400, jim wrote: > here at wkdu.org our minds are blown trying to figure out what our > problem is, we are using a dell optiplex gx1 with a soundblaster live > card. we are getting a nasty hum on the line in. the thing is we plug > a cd player directly into the soundcard the hum goes away and when we > plug the line in into anything else there is

Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux