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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

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

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

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

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 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

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

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 just downloaded the latest CVS. RTP streams from a X-Lite on a PC, to the Linux box running Asterisk and Linphone, seem to connect: > -- Called m4 > -- SIP/m4-dd8f is ringing > -- SIP/m4-dd8f answered SIP/m12-3649 > -- Attempting native bridge of SIP/m12-3649 and SIP/m4-dd8f But then Linphone sends an endless stream of complaints: > (linphone:20223):

Anyone know of anybody doing VXML with Asterisk and/or Linux? Tia Kevin

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", >

Does anyone know what this means? It is in DMESG, and we have people complaining about echo. Disabled echo canceller because of tone (rx) John

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

Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that