search for: eicke

Hi! I'm searching for a 1-800 number that simply plays music for a long time (>3mins) and no one picks up. I've bothered the AT&T lines so far when trying out my SIP->PSTN connection but then always someone answered :-) Anyone have a number? Christoph

--- Mike Mestnik <cheako911@yahoo.com> wrote: > Date: Fri, 25 Jun 2004 09:51:21 -0700 (PDT) > From: Mike Mestnik <cheako911@yahoo.com> > Subject: Re: IPP2P: Simular project l7-filter. > To: Eicke Friedrich <tady@gmx.net> > > --- Eicke Friedrich <tady@gmx.net> wrote: > > Mike Mestnik wrote: > > > http://sourceforge.net/projects/l7-filter/ > > > Providse and posibly replaces your project. Thay use regex(in > kernel > > > space) to filter...

Hello, Is there a way to route p2p traffic on a separate ISP connection, just as you would choose a separate connection for http traffic? I tried all sorts of setups based on: http://www.braindump.dk/en/wiki/?wikipage=PolicyRouting but with no luck. Please help :) (ipp2p is up and running) _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl

Hi! When I try to load the ztdummy driver via "insmod ztdummy", I get the following errors: /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_register I'm using zaptel-1.0.8 and

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all! I''m trying to shape traffic in a dorm''s network (4 mbit symmetrical internet link, about 200 computers, heavy p2p usage). The router is a p4xeon running linux 2.6.9 with the qnet patches (http://kem.p.lodz.pl/~peter/qnet/). When I activate ip_forward I get >20% packet loss and a lot of duplicates. Any ideas? I attach my shaping script. Thank you very much in advance,

I have a weird example which fails and I don't understand why. If I put dev.off() inside my function, it produces a bad image file. But if I run what I think should be equivalent code with dev.off() running outside my function, it works. Why does the location of dev.off matter? library(Cairo) library(ggplot2) foo<-function(){ Cairo(file="a.png") cat(dev.cur(),

Hi! I have two Asterisk installations, one being a 1.09 bristuff installation and one 1.08 installation. In the 1.09 installation I have the "Dial" command available on the CLI, in the 1.08 installation I don't. My question is now: was that a new feature in 1.09 or is it a bristuff specific thing or is my 1.08 installation simply lacking features? Thanks, Christoph

Hi! When I want to monitor the OSS/dsp channel through the Asterisk management interface, I get a "permission denied" error: Action: Monitor Monitor: OSS/dsp File: 1124096949 Mix: 1 Response: Error Message: Permission denied My permissions for /var/spool/asterisk look like this: pound:~# ls -la /var/spool/asterisk/ total 40 drwxr-xr-x 10 asterisk asterisk 4096 Aug 9 10:19 .

...ching support -debug option to print matchinfo into kernel log Please consult the README and documentation found at http://www.ipp2p.org for information about changes and new features. You can download the source tarball there as well and please give me as much feedback as possible. Kind regards, Eicke -- IPP2P - P2P traffic detection for iptables/netfilter More information at http://www.ipp2p.org _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Hi! I have a strange problem. In an AGI I tell Asterisk to playback a number, for example 31. I then use the AGI SAY NUMBER command and I only hear "thirty" and then get: -- Playing 'digits/30' (language 'de') Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable

Hi. I need a bit of help. I'm running Windows 7 and "out of the box" versions of Rtools and R (version 2.14.2) and am trying to compile a package that worked before. When I run the command: > Rcmd build -binary mypackage > Warning: unknown option '-binary' i get a warning and it creates the tar.gz file but not the zip file like before. Something has changed with

Hello, I guess a few of you know about the iptables-p2p project at the http://sourceforge.net/projects/iptables-p2p site. This suite has an excellent filter of which I use today and its running very smoothly. The main reason why I use this is because I desperately need to block torrents. IPP2P does not do this at this time (it seems to me in the doc at least). Is it maybe some way we could merge

hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group Protek-on: CaraMail met en oeuvre un nouveau Concept de S?curit? Globale - www.caramail.com

What happens if you take out the Zaptel I/F's? If it boots, you can correct whatever you did then replace them. hth -----Original Message----- From: Chris Earle (CBL) [mailto:cearle@cbltech.ca] Sent: Tuesday, February 28, 2006 7:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cannot boot machine up after working on zaptel.... Hi all, hard for me to explain this, but

Dear users, I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. Regards, Afzaal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050725/14265bb5/attachment.htm

I'm very interested in the redirect feature of Asterisk. So far I haven't gotten it to work. My scenario is that there is a two party call going on where I want to send one of those parties somewhere else. In the wiki is only an example how to send both parties to a meetme room. Is the ExtraChannel parameter required? This is what I have: Action: Redirect Channel: SIP/8080-e2a7

Is there an option txgain for SIP in Asterisk? My users all complain that their other parties think that they are way too silent even though they all have their mic volume all the way up and also enabled the 'mic boost' option. This happens with all the clients that we're using and also with different model headsets, so my last hope is txgain for SIP. Thanks Christoph

Hi! I've read in the archives that there are problems concerning Nikotel calls being disconnected after two minutes. I had the same problem yesterday. Is there a fix? There was only a "giving up" statement after the last e-mail in the archive, I'm about to do that too. Here's my sip.conf entry for Nikotel (left out the register stuff 'cause it's working):