similar to: loading misdn.conf strange error regarding out of range

Hi i've installed this software: ******************** SOFTWARE mISDN-1_1_7 mISDNuser-1_1_7 Asterisk-1.4.15 ******************** SOFTWARE misdn is correctly loaded by misdn-inist start Here there is the misdn.conf (copied from an existing and working installation with Asterisk 1.2.x and one BN8S0) ******************** MISDN.CONF [general] misdn_init=/etc/misdn-init.conf debug=0 bridging=no

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! 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 everyone! I'm sorry I can't be more specific. So here's the setup: a Samsung router with analog and ISDN ports. the phone company says the outgoing line is analog landline, but I'm sure it's some VOIP. so connected to the ISDN port of the router is a Fritz AVM card, used with mISDN. when I try to make a call with asterisk I get something like this: cli>>

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

Hello, I have a strange problem with misdn.conf When I run asterisk, I have this message : Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config: misdn.conf: "use_callingpres=yes" (section: general) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload". Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config:

Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the "misdn send digit" command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a

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,

Hello! Sorry, I'm sure it's stupid. but I've got a simple ISDN line and a simple ISDN-card, now finally running. :-) I'm using application Jack and asterisk (CLI) only to do my bidding. Now I can make calls. But how ca I setup my extensions.conf to receive a call? I've had an example like tis: [default] exten => 500,1,Answer() exten => 500,n,Jack() But it

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! 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! 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

I would like to know if the following misdn warnings are relevant. Currently, I don't need echotraining. However, I took a quick look at the * source code and l1watcher_timeout seems to be defined (echotraining was not found). Currently I'm setting l1watcher_timeout to 0 which is default (so I suppose that this warning won't affect me). Any comments on this? *CLI> misdn reload

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 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

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

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 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 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! 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?