similar to: Sip/IAX Clients for Linux

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

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

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

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 dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone

I am dying here with linphone (not sure if it is crap software or just me being an idiot) but out of the box debian installations of two linphones fail with a "Got SIP response 415 "Unsupported Media Type" back from 192.168.1.3" Can anybody recommend a particular SIP soft phone that broadly satisfies the following criteria? 1. Run on linux. 2. Simple to use and setup. 3. Is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i have a working Microsoft ISA firewall with buildin H.323 Gatekeeper.... So Far, i got registerd the asterisk on the M$ Gatekeeper... here is the h.323 configuration: ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 allow=all ; turns on all installed codecs dtmfmode=rfc2833 gatekeeper =

I'd like to cobble together a videophone from an analog phone, connected to an Asterisk FXS channel, and a co-located video camera, connected to a video grabber card on the Asterisk server (so I have a Linux video device providing the video stream). When a call is made from the phone, I'd like to somehow add the video and produce a SIP video call. I don't want to use any sort of

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 all! I'm newie to asterisk and I've been trying to make it work in order to use it with Linux softphones (H.323, SIP or IAX, I don't mind) and none hardware phone. I'm using asterisk packages from Debian SID (my distribution), asterisk, asterisk-config, asterisk-sounds, asterisk-h323. I've still not tried with any IAX softphone (gnophone?) but with linphone (SIP) I've

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, i tried to configure Gnomemeeting for Asterisk, because its, how it looks, the only tool which gifes me all i want for the use in linux... I have allready installed and running h323 support in asterisk and edited the h323.conf. But i have no chance to configure Gnomemeeting that it connects with Asterisk! I found also nothing useful in the

> http://www.speakfreely.org/ > > I think this would be one of the best real-world tests of the speex codec. > This software doesnt use ACM or directsound api's but uses straight C code. > I was thinking the speexenc/speexdec should be easy enough to add. The last time I looked at this it was still very much old news - mostly half duplex audio, does not adhere to any

Hi! I am newbie as Debian user as Shorewall and as GnomeMeeting. I try to configure Shorewall but i have still problem with GnomeMeeting. I have Debian Sarge, Gnome and Gnomemeeting, standalone computer and dsl internet. Thanks, Mitja

Hi all I have a problem using the Speex voice codecs when using GnomeMeeting on one side and NetMeeting on the other side. I use GnomeMeeting under Suse Linux 9.0 to communicate with a friend working under Windows XP and using NetMeeting 3.0. Under Windows XP / NetMeeting we have installed and registered the Speex voice codec. (You can find more information how we have registered the Speex codec

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

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

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

Hi all, I have seen many posts on the Shorewalllists dealing with H323. Although lots of them indicated that this is difficult process with kernelrecompilation etc. I just tried what seemed to be logical for me. Surprisingly it worked. Configuration: WS1 ----- FW ------ Internet ------- WS2/Shorewall WS1, FW and WS2 run Redhat9 with its standardkernel 2.4.20. FW and WS2 run Shorewall