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Hello, Can someone help me to understand the meaning of the following R line? list(fk5 ~ .) Thanks, Renata ----------------------------------------- Email sent from www.virginmedia.com/email Virus-checked using McAfee(R) Software and scanned for spam

Yes, I am a newbie too. I am having a problem with meetme. From what I have seen it will work without a Digium card but with audio problems. My goal is just to see how it works not the quality of the audio. When I dial into the conference room the following message is played: "That is not a valid conference number." On the console I get: "unable to open pseudo channel". As

Hi >Tdf bin TCC_TCA TCA_CR TCC_CR Time sn.rnc 117 117 258 27 314 (08/28/08 00:09:42) 50.21 118 118 251 30 291 (08/28/08 00:09:47) 50.21 119 119 247 28 289 (08/28/08 00:09:52) 50.21 120 120 251 29 282 (08/28/08 00:09:57) 50.21 121 121 276 39 320 (08/28/08 00:10:02)...

Hi, The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all my machines, no error, no log. Although, the CVS version works great on Linux. Is it anybody who knows how to compile it with mingw32 ? Or better, could anyone, who already has mingw32 installed, make a binary snapshot ? Thanks in advance, Jean-Christophe -------------- next part -------------- An HTML attachment was

Is anyone other than me having trouble dialing out via IAXTEL? I havn't changed my config files in weeks but seems that IAXTel calls (800 and FWD) stopped working in the past week sometime. Robert

I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup

I have my sip phones going into the context [from-sip] and would like to play an introduction message and then have the caller enter the extension. The message (dir-info was picked just to have something) doesn't play. Maybe I misunderstood the "s" extension. According to what I read it is executed everytime something enters the context. Obviously something was misunderstood. The

I am trying to compile * on SuSE 8.2. When doing the "make install" in /usr/src/zaptel I get the following error. ********************************************************** /usr/src/linux/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules freeIn file included from /usr/src/linux/include/linux/highmem.h:5, from

...s more specific >than "Re: Contents of Asterisk-Users digest..." > > >Today's Topics: > > 1. Re: Digium should develop and sell just Dummy card. For timing... (Adam Hart) > 2. Re: My Grandstream works, > but my X-Lite doesn't:no sound after 5sec (rnc Info Lists) > 3. Re: Wildcard TDM400P - FXO? (Steve Meyers) > 4. e100p in Australia (Stephen Dredge) > 5. Re: DISA and ringing tone (John Todd) > 6. Re: My Grandstream works, but my X-Lite doesn't: > no sound after 5sec (WipeOut) > 7. Re: Digium should develop...

Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro Some of the features are: - registering with Asterisk PBX; - can use any audio device as ring device (including PC speaker), independent of the play device;

Is anyone running Asterisk under Fedora Core 1 (http://fedora.redhat.com/)? If so, did everything with Asterisk work properly? I'm looking to migrate from Red Hat 8.0 to Fedora this week. Thanks.

On Thu, 4 Apr 2002, Akos Maroy wrote: > can you tell me more about these LADSPA plugins? LADSPA stands for Linux Audio Developer's Simple Plugin API (see http://www.ladspa.org/). Basically, it was pointed out on the linux audio dev (LAD) mailing list that numerous programs were using plugin architectures and all were different. So they fleshed out a plugin API and the rest, as they say,

Hello, I am wondering if someone has got any ideas that can help solve this problem. I have a dial plan that you call into, and depending on certain conditions it calls out on a number grabbed from a database. Something like this : exten => s,n,Do something exten => s,n,Do something else exten => s,n,Dial(ZAP/g1/${OUTBOUND},${timeout}) I need to log the time the person

Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a

Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things

Hello All, Thanks to those that responded to my problem of compiling on SUSE 8.2. I was not able to get the compile done so decided to put RedHat 9 on this system. After getting a RedHat supported NIC and RedHat installed, Asterisk compiled cleanly, one SIP phone is connected and voice mail works. No other tests have been run yet. A couple of days ago, Michael Farnworth asked about the smallest

Earlier today someone posted a RFC number related to voice mail. Unfortunatly I deleted the message so have lost the number and don't see it yet in Google. Can you please resend that to me? Thanks, Robert