similar to: Re: Connections, but no voice paths except by

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

Today's "frustrated programmer" award goes to Linphone, which has the following debug output: > (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that

Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, Daniel

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

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 am in the process of converting our company over to * to replace our ancient executone system. As part of the testing process my boss wants us to all run softphones on our desktops until he gets the phones ordered. Quite a few of us run fedora core 2, and I haven't had any luck getting a soft phone to work. Kphone works the best out of all I have tried but I get no sound out of

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, 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 There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark

Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just

I'm experiencing one-way voice paths, followed by a hangup on one softphoine and not the other. Also, caller ID has odd outputs -- and I wonder if the problems are related. My configuration has Asterisk and a Linphone softphone running on the same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect to the Linphone instance. When I call from the PC to Linphone: * I call

Surely there has to be one soft phone that works under Linux. I've tried: kphone - it sometimes complains about the need to release the sound device linphone - sssssssslllllllllooooooooowwwwwww iaxcomm - needs some strange widgets various others - either only supplied as binaries, or just plain don't work, or won't compile. Is there just one out there that is guaranteed to work with

Hi, I'm trying to set up asterisk, and I'm having some problems with a simple register. I'm not sure where to start even -- It seems that the problem is with the response to the digest authentication, but I'm not sure how to fix that. The log below is from linphone, but I see the exact same thing with kphone and xten from a indows box as well. I've tried changing the realm

I have a software-only PBX set up. I can register various softphones and they will call each other -- but I've never succeeded in getting any voice routed from any of the softphones. Only the console will transmit audio. I am writing to ask if I have missed some obvious step in configuring the system. Conditions: (1) Softphones running on the same machine as the PBX: Only Kphone seems

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]

For what it is worth, here are my experiences. When I first got my zaurus I compiled R version 1.4.0 for the it. It took some time to do, but managed with the skiffclusters at familiar.org. (libf2c was the hardest part). Compilation should be easier now that there are onboard compilers but I haven't tried. I thought this would be neat to have in the classroom. (It did get one student to buy

https://bugs.freedesktop.org/show_bug.cgi?id=70130 Priority: medium Bug ID: 70130 Assignee: nouveau at lists.freedesktop.org Summary: unable to compile fragment shader program Severity: normal Classification: Unclassified OS: Linux (All) Reporter: infyquest at gmail.com Hardware: x86 (IA32)

Hi, We have set up our Asterisk server, our extension.conf and sip.conf according to http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=4 It's quite basic, and extension.conf is set up properly. The difficulty we are now encountering is in sip.conf, in trying to get any softphone to register at our own Asterisk server. We have searched the mailing list, and find bits and

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

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 >>>