similar to: 2003-06-10 CVS: softphone connection failures

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

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

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

Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. I've been chasing ALSA/Asterisk/client problems in one form or another for some time now. In previous versions of Asterisk and ALSA -- i.e., last week -- I could load either chan_oss.so or

Hello, I'm trying to compile linphone 3.0.0 in Ubuntu And I'm getting the following compilation error. I somewhere read that the fault happens because of the gcc 4.3.2 that debian based linux uses. Is there any solution (except downgrading gcc) Thank you in advance if /bin/bash ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I.. -I../include/ -I.. -I../../oRTP/include

I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I

strace on file access in asterisk shows that * is not even attempting to access the voice files. If I *manually* load app_playback.so, app_macro.so, and then pbx_config.so, I they will load and I get a dialplan. Ok, that's a problem -- autoconf is clearly not working, or there's some other related issue. So I try to use the demo and do "dial 500". This should connect and

Sorry 'bout that "vacation" message. Procmail usually is smart enough to avoid sending replies to mailing lists. I put in a rule to prevent this from happening again. -- Moshe Yudkowsky * http://www.Disaggregate.com

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

strace does show that that modules.conf loads: > open("/etc/asterisk/modules.conf", O_RDONLY) = 8 And that I do get some of the channels loading, e.g., the modem channel: > open("/usr/local/lib/asterisk/modules/chan_modem.so", O_RDONLY) = 8 And if I load the apps via "load app_playback.so", >

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

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

Hi, I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? Thanks in

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

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

Hi folks last weekend, I tried Windows Messenger first time and was stunned by the little latency it gives. Until now, I've been using softphones on Linux exclusively, like iaxcomm, linphone and sjphone, and they all give me about 1, at times even 2 secs delay. Whereas Messenger really seems to be in the millisec range. Of course, I'm now curious why there is that difference. Clearly,

Hi guys, Being a true Linux geek, I've never been too much into sounds or sound files other than a few .mp3 songs I got. My question is pretty straightforward and simple. I see that the music format of choice for asterisk is .gsm. What can I use to listen to files in .gsm format and what is the most effective way of recording files into .gsm format? The last part of the question is

--T4sUOijqQbZv57TR Content-Type: text/plain; charset=us-ascii Content-Disposition: inline Dear R, I would like to report what I think is a bug in R. I am running R within emacs on a Digital AlphaStation. See the version information at the end of my R session for details. I also attach a copy of the file that is read in the `read.table' command. Here's my R session, with a few

dear expeRts, i would like to merge the data frame entries in a list. for example: > #input > myl <- list(q1=data.frame(id=c("Alice", "Bob"), grade=c(90, 49)), q2=data.frame(id=c("Alice", "Chuck"), grade=c(70, 93)), q3=data.frame(id=c("Bob", "Chuck"), grade=c(84, 40))) > #output > (mydf <-

Moshe, I was having the same problem with my software only asterisk pbx setup. I was using two kphones on different machines, connecting through a machine running asterisk. They would connect just fine, but voice was not getting routed through. I installed linphone, which can be found at wwww.linphone.org, on the machines running kphone, and can now get very clear voice communications,