search for: anrufe

I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. Thanks, Brian

...9;t know how to pre-handle message type SEGMENT (0x60) | My Config files: extensions.conf CODE:SELECT ALL <http://forums.digium.com/viewtopic.php?f=1&t=77155&sid=3679c2a13cbbf9aa2e75d02a2f00c8b3#> |[general] static=yes writeprotect=no [isdn] ; Ankommende anrufe exten => 833762,1,Dial(SIP/3762,45,r) ; Rausgehende Anrufe exten => _0[1-9].,1,Dial(DAHDI/g1/${EXTEN:1}) [default] include => isdn | /etc/dahdi/system.conf CODE:SELECT ALL <http://forums.digium.com/viewtopic.php?f=1&t=77155&sid=3679c2a13cbbf9aa2e75...

Dear friends of R, I want to produce a pdf file with the contents of a matrix. I employ the latex command in combination with dvi, both contained in the Hmisc package. It seems to me that the function does not correctly set the directory. > tbl.loc <- matrix(1:4, nc=2) > latex.obj <- latex(tbl.loc) > dvi(latex.obj) warning: extra args ignored after 'cd' H:\PROJECTS\data

Hello, I have a problem with our asterisk server. (Version 1.4.15 RPM Build running on Fedora Core 7) After 3 day's running, i'v got this message in my logfile: Dec 14 13:14:09 amsec-tk1 asterisk[4321]: VERBOSE[11363]: -- Executing [9020 at AMS-UPPORT-HOTLINE:1] System("SIP/SN2400-67ec6aa0", "echo -e "Eingehender Support Anruf am 14.12.2007 um 13:14 durch die Nummer

...hone which sets its >CallerID to 1500. > >Can anyone be so kind to tell me what is shown to the callee in either >case? > I can only tell you, that after I set exten => _0[23456789].,1,SetCallerPres(prohib) the callees phone displayed "Caller" (or in German "Anrufer"), which is just what I wanted. Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de ***************************...

...type = MixMonitor joinempty=strict leavewhenempty=strict maxlen=20 member => Agent/21 member => Agent/22 member => Agent/23 The agents login with AgentCallbacklogin wiht AgentCallBackLogin(21||*21 at agents); agents-context in extension.ael looks like this: context agents { // Anrufe von queue an agenten *21 => &queuecallagent(${EXTEN:1}); *22 => &queuecallagent(${EXTEN:1}); *23 => &queuecallagent(${EXTEN:1}); } macro queuecallagent(agentnr) { Dial(IAX2/${agentnr}||xtTg); switch(${DIALSTATUS})...

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...

Deutsche ?bersetzung folgt / German version following ===================================================== Hello, i have Asterisk running with 2 ISDN-Cards. One AVM Fritz for connection to german ISDN and one HFC-compatible-Card (NT mode) for connection to ISDN-Phone (later: ISDN-PBX). Here is my actual installation: ISDN -> Fritz - ASTERISK ? HFC-NT <- ISDN-Telephone If i pick up my

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. Ich h?tte es aber gern so das ich erstmal den H?rer abnehmen kann und dann...