similar to: Problem with BRI_STUF / direct connected ISDN-Phone

Displaying 20 results from an estimated 110 matches similar to: "Problem with BRI_STUF / direct connected ISDN-Phone"

2004 May 08
3
asterisk with german SIPGATE ?
hi anybody running with german SIPGATE? my configuration don't works :-( regards thorsten@gehrig.de
2004 May 11
1
Need help: X100P connection/configuration in GERMANY
hi i need help in connecting a X100P-clone in germany. Basic questions: a)what configuration do I need ? what is the difference (maybe explain in german? via email?) fxs_ls: FXS (Loop Start) fxs_gs: FXS (Ground Start) fxs_ks: FXS (Kewl Start) b) can I use one card for connecting to the telephone network (in this case my analog pbx) and the same card for connecting an analog phone
2005 Jul 20
1
how to define a port range?
hi, i´am new in tcc (tcng). i try to define my qos for VoIP-Services. For this i wantto define a class for a port range 10000 till 15000. how is the right way? this down works: class (<$voip>) if tcp_sport => 10000 || tcp_sport <= 10000 ; are there any examples of real installations - maybe including VoIP, HTTP and P2P services? regards thorsten gehrig
2004 May 06
3
Dial internal phones problem - zaphfc
Sorry that I wrote in german : Ich benutze asterisk mit dem zaphfc Treiber. Jetzt hab ich folgendes Problem, habe 2 ISDN-Telefone angeschlossen. zaphfc im nt-mode. Anrufe von ausserhalb per sip (sipgate.de) kommen an. Wenn ich aber intern zwischen den zwei Telefonen (Ascom Eurit 30) sprechen m?chte geht das nur wie folgt : Erst die Nebenstelle w?hlen und dann den H?rer am Telefon abnehmen.
2004 Sep 10
1
support for more than 8 channels
Hi, we have a speech recognizer here that gets some of its input from a microphone array. To the point we used shorten for the archiving of the material, but since we want to change the license of the speech recognizer to gpl we are searching for an gpl alternative (like FLAC). It seems to be pretty good for our needs (24bit rather than 16bit in shorten and so forth). The only problem is that our
2013 Sep 10
1
Sieve Filter global vs user specific
Hi at all! I'm actually fighting to make sieve in dovecot work and made quite a success by now. However, I still fail at the following constellation: Background: I'm a Mail Admin of a small IT department and we are already using Dovecot as LDA with a filtering server. Any user can easily create filter rules that apply to him (to make it easier for my colleagues we use the Roundcube plugin
2005 Feb 13
2
3.0.11/MirOS password change problem
Hi! Does this sound familiar, before I try to look deeper into it? [2005/02/13 20:10:16, 0] /usr/ports/net/samba/w-samba-3.0.11/samba-3.0.11/source/libsmb/smbencrypt.c:decode_pw_buffer(539) decode_pw_buffer: incorrect password length (1251354155).
2010 Jan 18
0
LinuxTag
Hey theora people, I would like to announce the call for paper for the LinuxTag 2010. This year, video got it's own track ( 2x3 hours! ). I would love to have some theora specific presentations on the LinuxTag. So please send your paper abstract and/or join us in june in Berlin! -Yorn (Member of the LinuxTag Program Comittee) --- Here is the official announcement with detailed
2011 Jun 28
1
Samba auf neuem Datei-M2
Hallo Herr Jahn, falls Sie eben Anrufe von uns bekommen haben und Sie nichts geh?rt haben - das liegt an Problemen, die wir momentan mit unserer Telefonanlage haben. Deswegen kurz per Mail: Ich habe die Pakete auf Datei-m2 installiert. Da wir mit dem Setup von den Samba-Servern ja zuletzt einige ?berraschungen erlebt hatten, w?rde ich vorschlagen, dass Sie den Server erstmal mit einem anderen
2011 Mar 16
0
Setting up 1.6.2.9 on Debian 6.0 Squeeze
Hello. I would need some help trying to setup Asterisk 1.6.2.9-2+squeeze1 on a Debian 6.0 system. I'd like to use the Debian packages, hence the "strange" version number? Since I'm new to Asterisk, I'm trying to follow "The Asterisk Book" at http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.html and created a VERY basic sip.conf; see
2007 Jun 22
2
asterisk 0 dial outgoing call
Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give
2013 Apr 18
1
parSapply can't find function
Here is the code, assuming 8 cores in the cpu. library('modeest') library('snow') cl = makeCluster(rep('localhost', 8), 'SOCK') x = vector(length=50) x = sapply(x, function(i) i=sample(c(1,0), 1)) pastK = function(n, x, k) { if (n>k) { return(x[(n-k):(n-1)]) } else {return(NA)} } predR = function(x, k) { pastList = lapply(1:length(x), function(n)
2007 Jan 11
1
nut-upsd on mac os
Hello. I have an cheap seriell to usb adapter with the Prolific chipset hooked on my mac, but I don't know which tty to configure. OS X regognizes the device in its system profiler. /dev/ttys0 seems wrong. Can someone please enlighten me? Thank you for your help! Greetings, Robert Welz
2007 Aug 21
1
Problems with overlap dial and Xorcom Astribank BRI
I have a strange problem with overlap dialing. I installed an asterisk server between a Siemens HiCom PBX and our telephony provider. Everything is working fine except some strange problems with the dialing of the fax (connected to the HiCom PBX). It seems to me that if dialing takes too long Asterisk just hangs up the channel without recognizing that the fax machine is still dialing: (Fax gets
2009 Jan 25
2
how to build a small asterisk pbx
Hi i must build a small phone pbx system. My friend has : 3 phone analog lines 6 phone extension How can i build that thelephone system? Nightduke
2006 Jun 10
2
Outside mails
Hi, I recently found out that I was able to get system mails, but not mails from the outside world. What setting do I have to fiddle with for this? Regards,
2008 Jan 12
0
libfishsound 0.9.0 Release
FishSound 0.9.0 Release ----------------------- libfishsound provides a simple programming interface for decoding and encoding audio data using Xiph.Org codecs (FLAC, Speex and Vorbis). This release is available as a source tarball at: http://www.annodex.net/software/libfishsound/download/libfishsound-0.9.0.tar.gz New in this release ------------------- This release introduces support for
2008 Jan 12
0
libfishsound 0.9.0 Release
FishSound 0.9.0 Release ----------------------- libfishsound provides a simple programming interface for decoding and encoding audio data using Xiph.Org codecs (FLAC, Speex and Vorbis). This release is available as a source tarball at: http://www.annodex.net/software/libfishsound/download/libfishsound-0.9.0.tar.gz New in this release ------------------- This release introduces support for
2005 Feb 06
0
liboggz 0.8.6 Release
Oggz 0.8.6 Release ------------------ liboggz is a C library providing a simple programming interface for reading and writing Ogg files and streams. Ogg is an interleaving data container developed by Monty at Xiph.Org, originally to support the Ogg Vorbis audio format. This release is available as a source tarball at: http://www.annodex.net/software/liboggz/download/liboggz-0.8.6.tar.gz New in
2005 Feb 06
0
liboggz 0.8.6 Release
Oggz 0.8.6 Release ------------------ liboggz is a C library providing a simple programming interface for reading and writing Ogg files and streams. Ogg is an interleaving data container developed by Monty at Xiph.Org, originally to support the Ogg Vorbis audio format. This release is available as a source tarball at: http://www.annodex.net/software/liboggz/download/liboggz-0.8.6.tar.gz New in