search for: kphone

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] exten=&...

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....

-------- 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...

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...

I'm having trouble with kphone on our system. It's using ulaw on an internal network. No NAT. I had it working fine with very similar hardware (an old Dell Optiplex GX1) running as an LTSP terminal. But then I put the same sound card in an Optiplex G1. Kphone will answer the line fine when I call it (call coming from th...

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 e...

i tried asterisk ilbc codec against kphone. when the call got connected, i started to immediately get these kind of message to the console: WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_fr...

Le 25/01/2023 à 17:56, Antony Stone a écrit : > On Wednesday 25 January 2023 at 16:46:14, Daniel wrote: > > On Sunday 01 January 2023 at 17:30:03, Antony Stone wrote: > >>> The [globals] section of that dialplan includes: >>> >>> Kphones=SIP/KC470IP&SIP/KSnom870 >>> Sphones=SIP/SYealinkT38G&SIP/SGC610IP >>> Allphones=${Kphones}&${Sphones} >>> >>> On the new system, the variable Allphones ends up containing: >>> >>> ${Kphones}&${Sphones} >> >> I do...

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 uses my asterisk...

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 1.0.7 and 1.0.9 asterisk versions. Asterisk does not c...

...om 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 follows. ** Linphone: Check out my SRPM at http://www.bigu.org/SRPMS/ Sound is fine. Doesn't seem to pick up anything from the microphone, though. ** kphone: Check out my SRPM at http://www.bigu.org/SRPMS/ Sound is fine. Picks up sound from the microphone, but the echo-test repeats it back after passing it through a Mr-Roboto filter. ** tkPhone: Sound is fine. Doesn't seem to be reading from the mic, no traffic going over the network after...

All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is trying...

Im new to Astererisk. I compiled the latest CVS and setup the server. It looks like things are working. I'm running kphone, x-lite and sjphone to test things out. The kphone (local to the asterisk server) can call and receive calls from any of the 2 windows machines. The first windows phone I start I can send/receve calls the second one I cannot. I. No matter which one I start first only the first one works. The l...

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454 -------------------------------- REGISTER sip:192.168.100.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.13;rport CSeq: 4399 REGISTER To: "sjphone2" <sip:sjphone2@192.168.100.3> Expires: 900 From: &qu...

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 softphones. Thanks much for sharing your expe...

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 formdomain=192.168.6.244 .... .... [kphone] type=friend ho...

Hi. I've just bought SIP telephony service from a Swedish telco. I've managed to make and receive calls with kphone. Now I want to set up asterisk to be able to add fancy features like voice mail and recording conversations. But first I have to get the basic setup right. I'm running asterisk and kphone on the same machine, behind at NAT-router. When I make a call (from my regular phone) to the SIP-number I...

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 extension) with my kphone also fail. exam...

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 [ro...

Hi. I have a very old dialplan (ie: a dialplan for a very old version of Asterisk) which I've just transferred to Asterisk 16.28.0 The [globals] section of that dialplan includes: Kphones=SIP/KC470IP&SIP/KSnom870 Sphones=SIP/SYealinkT38G&SIP/SGC610IP Allphones=${Kphones}&${Sphones} In the old system, this results in ${Allphones} containing: SIP/KC470IP&SIP/KSnom870&SIP/SYealinkT38G&SIP/SGC610IP I can use this in a dial() command. On the new system, th...