similar to: gtalk/jingle full report

Hi! I just tried to call a friend using jingle, but I got refused. Errorcode was 502, he tried to call me, heard it ringing once and then it stopped. I used: originate jingle/gtalk_account/friend at jabber.linuxlovers.at [application] I'm registered to googletalk, but this should mean no harm, or should it. Once I was able to receive a text-message from him, but couldn't

Hello everyone! I just installed the latest asterisk from svn. Now I'm retrying my luck with gtalk and jingle. I have moved so my basic setup has changed a bit... I'm not sure if it helps or hurts. I tried this: call myself: channel originate gtalk/gtalk_account/juliencoder at googlemail.com application \ Jack i(system:playback_1)o(system:capture_1) I got some notes about a lot

Hello everyone! So I've just scanned through the debug log, defined like this in logger.conf: full => notice,warning,error,debug,verbose I couldn't see any reason for the connection not working. I called my friend, he heard ringing, accepted the call and then it got hungup. I didn't see any output from app_jack though. Any idea, how I can get more output from app_jack?

Hello everyone! This is my problem: I try to do gtalk, but my asterisk server uses the local IP 127.0.0.1 or perhaps the 192.168.*.*. Now I've heard, that a NAT router can help there. I was told it's the way the windows-world does the trick, when they sit behind a router/phonebox/modem. Does anyone know a good software that will do the trick on Linux? I'm running Debian Lenny

Hi everyone! I couldn't find anything expressive about gtalk dialstrings. It doesn't seem to work. I'm not sure why, so I'll start at the easiest point. The syntax I found was: gtalk/my_account_name/buddys_account_name at gmail.com Is this correct? And does any of you googletalkers know, if a simple google-mail account is enough to use the talking bit, or do I have to

Hello everyone! So I hacked app_jack.c today, as best I could. Whic came mostly down to inserting ast_log() messages. I discovered the following with JACK: When it starts, it tries to read 512 bytes and only gets 0. That clears up after a while. Sometimes a good time later than the reading comes the writing. And there the real strangeness might begin. Because usually the framebuffer

Hello everyone! So I tried to test gtalk with a friend. We could both see each other. He uses the gtalk application for Windows. So I tried to call him and he got a ringtone. But when he picked up, he got a missed. When he called me, he got a dial tone and then after one "ring" he got a woman saying: "Sorry, the person your are calling is not available. Please leave a

Hello everyone! Would someone be willing to test googletalk with me? Please reply in private, I'll get you my account and we can exchange anything else, that's necessary. Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide

Telepathy client with Gtalk support, so we should be able to call him soon :) Also please file your bug report on the bug tracker : http://bugs.digium.com Thanks! Philippe On Tue, Oct 28, 2008 at 12:41 AM, Julien Claassen <julien at c-lab.de> wrote: > Hello everyone! > Philippe, you told me to make a bugreport. Well, here it comes, I'm still > not sure, if tis is a bug or

Hello everyone! I'm just trying to set up my new asterisk (version 1.6.2.8-rc1). I'd be very grateful, if someone could help me here. I'd be very glad, if one of you could test googletalk with me. Last time I tried (in 1.6.0.x times) it wouldn't work in the end. But here are my gtalk and jabber.conf files. Could you please take a look and tell me, if the settings same sane?

Hello everyone! I've just switched from Asterisk 1.6.0-beta9 to 1.6.0.1 and my mISDN is not working. Here's what happens, if I try to call the line: bach >> P[ 1] --> !! lib: No free channel! P[ 1] --> we have already send Release_complete I haven't changed the configuration fles. Should I change something there? If you need more info, just tell me and I'll

Hello everyone! Perhaps I missed something: But where can one download the beta-version of the new asterisk skype channel? Can it work with 1.6.0-beta9? I tried to browse the digium downloads, but it's dificult, if you're blind and only have a text-based (almost no javscript) browser. Thanks for any good hints and pointers! Kindest regards Julien -------- Music

Hello everyone! Sorry, if the whole task is silly, I'm new to this. I have my newly installed asterisk (1.6.0-beta9) and my AVM Fritz a1 card. I have a simple German isdn line and I have a microphone, headphones and a running JACKd (JACK Aduio Connection Kit). The question: Can I (mis)use my asterisk CLI interface to make and recieve calls coming in/going out via the ISDN-card,

Greetings! I now found someone to test gtalk with and found out, that app_jack has a problem here. My voice gets transmitted fine, but I only get white noise from the other party. I tried to set my JACK samplerate to 8000 to make sure it's no libresample problem, the results were the same. My setup is: Linux Debian Lenny Kernel: 2.6.30.4 PREEMPT (self-built) JACKd: jackd version

Hello everyone! I haven't seen anything new about the JACK support in Asterisk and I was wondering, if anyone has experience with a current release of Asterisk, JACK and mISDN/googletalk etc. I'm thinking of installing a new version (havingcurrently 1.60-beta9. But the excercise would be pointless, if it doesn't help. Kindly yours Julien -------- Music was my

Hello! I'm still wondering, how to park a call with an ISDN line. The setup is the asterisk server only, controlled via the CLI. I can originate a call and I can tell asterisk to start the JACK application. But I can't then park the call. I tried it with sending DTMFs with misdn send digit, no luck. I had a look at the CLI, but didn't stumble upon a command to park the call.

Hello everyone! I still am not much further along with my sip calling. I changed my sip.conf taking suggestions from the net (voip-info.org in particular). I changed iptel's position from friend to peer. I turned on and off nat, I chose different codecs in first place, entered my outward IP as fromdomain and uncommented the register directive with correct values. All I get is two

hello everyone! I just got the newest asterisk SVN: trunk# svnversion 152803 and compiled it. then I made some test-calls. 1. Calling my mailbox. It worked, but quality was not good, in comparison to 1.6.0-beta9. I called via mISDn. 2. Just call myself. Result: Ringing and asterisk segfaulting. 3. Same for calling some gtalk-number and using app Dial mISDN/1/my_number. I got this a

Godd evening! What happens if someone calls and asterisk doesn't "Answer()" itself, but another analog phone does? Can I somehow catch this situation in my dialplan. I have an ISDN line, but with it I got a box with an adapter for good old analog phones. This doesn't seems to be directly connected to the ISDN line asterisk sees. But somehow, it must know, that the call

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I