similar to: Still sipping frustration - only getting state ACK

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

Hi, is there anybody who knows how to force Asterisk 1.4 to use soxmix instead of sox? Thank you. Giorgio

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

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! 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

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! 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! 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! 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?

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

Hi, I'm trying to run Asterisk 1.4.23.1 on a small ARM linux board (TS-7800). Everything compiles fine, but on startup Asterisk always crashes while loading chan_sip. If chan_sip is removed, it starts up fine, but I really need SIP to work. Any ideas? Thanks. -- James

Hi, everyone I'm investigating if I could give asterisk CLI access to one of our clients. If I add that user to asterisk group and set his shell to /usr/sbin/rasterisk, is there a possibility for a user to brake our of asterisk CLI to normal shell? Thanks in advance

Hi, URL http://lists.beronet.com/mailman/listinfo/misdn-asterisk in http://www.misdn.org/index.php/Support page returns Not Found. Is this list still active ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090107/9ef186d4/attachment.htm

Hello I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine. I am using the OpenVox B200P ISDN card. My problem is that even though chan_misdn module seems to be loaded correctly with Asterisk (I can see it using 'module show' command) the misdn commands are not available to me in the CLI so I cannot tell if my box is correctly interfacing with the ISDN card Any ideas

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

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