similar to: Slow wiki?

Hi Everyone. Some time ago, I was told that it's possible to implement an incoming Fax server with extensions using only one PSTN line, like this: PSTN number: 1234567, that's the line connected, FAX numbers like: 1234567-00, 1234567-01, 1234567-02 and so on are routed to their respective recipients. I do not want real DID (which would mean buying a bunch of numbers from the telco,

I'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The

Our main US mirror is cran.mirrors.pair.com, AKA cran.us.r-project.org. Pair.com keeps statistics on traffic on the mirror sites, and I got all excited when I looked at this page: http://mirrors.pair.com/pair/stats.html and saw that CRAN was 5th most popular over the last month, getting more visitors than Apache, MySQL, OpenOffice, etc. Then I looked at this graph:

I guess this is mostly for Curt, but I wanted to toss it out there. I could really use a wiki to capture the information that gets posted to this list. Eventually we''ll want FAQ''s and HOWTO''s, as well as a "real" manual, and the wiki is great for collecting raw material. Also, we should have links to related projects, like wxrbbr, and projects written in

Are there any IAX speaking "hardphones" out there? If so, can anyone offer comment on their quality? Thanks! -Dorn

I have a UIP200 on the way for eval. Does anyone have tips or tricks to get it working right away with * ? I hate having to go through the pain someone else braver than I went through already. :) Tim --

I'm looking into solutions for providing a live stream of an event in Belgium [1] - for example, as follows: * Event --> mobile phone --> software answering machine --> Internet server * Event --> mobile phone --> VOIP --> Internet server The live stream should be available in a format so that people can listen to it using XMMS or similar software. Comments? Would

As I've been reviewing the LLVM documentation, I noticed several typos or things that could be explained in more detail. My compulsion was to just go and fix it, but I can't do that because I have no access to the sources. Has there been any thought of putting up a Tiki Wiki for LLVM? If you're not familiar with Tiki Wiki, its a PHP/MySQL based content management system.

I am interested to learn if I need to have ztdummy installed if I do not have any zaptel hardware in my machine? I have found a lot of references with RTP problems which were related to RTP timing (or lack of it). My problem is that voice coming from SIP hardware is OK, but voice going from asterisk to SIP hardware is choppy, full of noise or completely cut-off. Am I going to solve my problem

I have selected 4 software packages for us to evaluate in order to decide on the best possible engine for the much-requested cAos Community Wiki. All 4 samples are now up and running for you to try out, play around with, and evalute. The URL's are as follows: https://caos.nplus1.net/c-arbre/ https://caos.nplus1.net/dokuwiki/ https://caos.nplus1.net/pwp/ https://caos.nplus1.net/tikiwiki/

Is anyone out there using app_dbodbc (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it? I was able to get it all working, but it causes * to segfault every now and then. It does not appear to be related to any specific function (ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I noload the module, everything works fine, but when its running, after calls to any of the

http://www.voip-info.org gives: Warning: mysql error: No Database Selected in query: select `name` ,`value` from `tiki_preferences` in /var/www/html/tikiwiki-1.8.2/lib/tikidblib.php on line 133 Values: Array ( ) $result is false $result is empty Was going to grab a link to give to Florent regarding his CTI thread and question about how to program against the Asterisk API...

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Can the Dial() command tell the difference between busy and congestion? At the moment it seems to be treating them both the same on my server. I want to route the calls out via a SIP gateway unless that is congested, in which case dial out through my POTS line (using an X100P). It seems a bit pointless to try dialling the POTS line when the SIP

Hi, is the following possible? I would like to transfer a call to my "personal" MeetMe conference room and get transferred there automatically as well. Currently I can transfer the call to the conference, have to hangup and then call the conference number myself. I would love to have this in one quick function. Moreover is there a way to disable the "You are currently the only

hello is there a gui that would allow me to configure everything from phones, to extentions, to voice mail to basicly everything that asterisk can do? I did go to www.voip-info.org and none of the guis I saw there do the trick and the ones that come close aren't downloadable just wanted to see status on this thanks hank ---------------------------------------- My Inbox is protected by

I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I

Hi there, here is my attempt to initiate a "restart when convenient" from a software SIP phone. exten => 588,1,Answer exten => 588,2,Wait(1) exten => 588,3,Playback(restart-convenient) exten => 588,4,Wait(1) exten => 588,5,Authenticate(00000) exten => 588,6,System(/usr/sbin/asterisk -rx "restart when convenient") exten => 588,7,Hangup The problem: We

So far in January, we've had 726 messages on -users. December 2003: 2.978 messages November 2003: 3.410 messages October 2003: 3.177 messages December 2002: 741 messages December 2001: 67 messages ...the project is growing. /Olle

Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? Thanks, - Darren