search for: kphones

...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=>1,1,Dial(SIP/jitha,20,tr) exten=>2,1,Dial(SIP/sudhananda,20,tr) Both the Kphones got registered to the asterisk but when i dial the number it gives me the following log on asterisk Asterisk Ready. *CLI> -- Registered SIP 'sudhananda' at 172.16.2.35 port 5060 expires 900 -- Executing Dial("SIP/sudhananda-aa77", "SIP/jitha|20|tr"...

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

-------- 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'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 the * machine), but when we try to get kphone to dial, each GUI

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 =>

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_framein): Huh? An ilbc frame that isn't a multiple of

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

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

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

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

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

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

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

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

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

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

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, the...