search for: anrufer

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

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

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 ****************************...

Hi all! My problem is that calls being placed in the queue, and are waiting while the agents are busy, when an agents is then free they gets connected to the agent but there is silence (no voice). If a caller has not to wait in the queue, there is no problem. My agents have an iax2 client, and imcoming calls are over SIP. queue.conf: persistentmembers=yes autofill=yes ringinuse=no

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

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.