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Hallo Leute, ich m?chte gerne die Samba Doku ins deutsch ?bersetzen. Ich arbeite schon lange mit Samba und denke es w?re kein ?berm?ssiger Aufwand. Da ich Samba aber meist als Pdc benutze bin ich mit einigen Features technisch nicht so vertraut. ( samba und ldap an einem win server etc ) Ich w?rde mich freuen wenn einige Deutsche mitmachen w?rden damit w...

Hi list! I think there are many german users in this ML, that use Asterisk with the new line of Deutsche Telekom (Magenta Zuhause). My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right now I can just hope, that I configured my Asterisk well to work with Deutsche Telekom, but I cannot be sure, since I can't test it... So my question: can someone using Asterisk with Deutsc...

Hi list! I already had the problem last year, then it would be solved (surely from some technician by Deutsche Telekom on their servers), and now I have the problem again (but I didn't changed my Asterisk configuration). The problem: after 15 minutes will the call dropped, but only if the call is to another nation! If I just call another phone in Germany, I can speak longer than 15 minutes... Someone...

Am 14.06.2020 um 17:05 schrieb Antony Stone: Hi Antony, > You mean that the Thomson phone is registering to Deutsche Telekom? > > I thought it was registering to your Asterisk server. Sorry, I didn't read correctly your test 2b... Normally my Thomson phone is registering to my Asterisk server. I tried to register the Thomson phone directly to Telekom's server, to check if the problem could be in...

...ot;) > > RMLANG <- data.frame(RM_EN,RM_DE,RM_RU,RM_CN,RM_JP,RM_KR) > nrm <- NROW(RMLANG) > > for(i in 1:nrm) + { + cat(format("English", width = 12, justify = c("left")), as.character(RMLANG$RM_EN[i]),"\n",sep="") + cat(format("Deutsch", width = 12, justify = c("left")), as.character(RMLANG$RM_DE[i]),"\n",sep="") + cat(format("Russian", width = 12, justify = c("left")), as.character(RMLANG$RM_RU[i]),"\n",sep="") + cat(format("Japanese",...

jg <webaccounts173 at jgoettgens.de> schrieb: > It doesn't really depend on your sip.conf and Asterisk. Your gateway/router > will be the major problem. My summer project will be to look at session Are you sure? Right now I'm using an italian SIP-Provider (Messagenet), configured in my sip.conf and I can receive calls without any problem... So, I don't think, I have to

...tml thx for his work to Mr Stefan G. Weichinger mailto robowarp@gmx.de if you want to be a part of the Translation Team you can view other files translation progress and the readme beyond http://www.robowarp.de/smb/trans/ Best Regards Robert R?gner Hallo, die erste Datei der Samba Doku wurde ins Deutsche ?bersetzt http://www.robowarp.de/smb/trans/net.8.html Vielen Dank daf?r an Stefan G. Weichinger falls ihr Euch beteiligen wollt mailt an robowarp@gmx.de Ihr k?nnt den Fortschritt der ?bersetzung anderer Dateien unter http://www.robowarp.de/smb/trans/ einsehen MfG Robert R?gner

> I think there are many german users in this ML, that use Asterisk with the > new line of Deutsche Telekom (Magenta Zuhause). > > My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right > now I can just hope, that I configured my Asterisk well to work with Deutsche > Telekom, but I cannot be sure, since I can't test it... > > So my question: can someon...

Hallo wer will PXE Linux in deutsch haben? Ich hatte mir mal die M?he gemacht es zu ?bersetzten.

...yet again. I'm sure you can get the Thomson to connect to Asterisk via > some external network, since you say you can do this from your Android phone. > Again, check the call quality. I tried it on the network of a friend. Not possible to establish a connection at all... I *suppose* Deutsche Telekom just allow a logon on their servers from the IP of the user, who tries to log on (with other words: my VoIP login can just log on from my current IP)... This would explain why I didn't got my mobile phone connecting to the Telekom's server and establish a call... I also tried to...

Hi! My telco is Deutsche Telekom and they got about 30 SIP servers right now. Currently I've set up a template for incoming calls in sip.conf and added each SIP server by it's IP address like this: [DTAG-in-1](DTAG-in-template) host=217.0.16.103 ... [DTAG-in-30](DTAG-in-template) host=217.0.20.99 I've do...

Package: logcheck Version: 1.2.62 Severity: minor # Send the results as attachment or not. # 0=not as attachment; 1=as attachment # Default ist 0 ^^^ MAILASATTACH=0 -- System Information: Debian Release: lenny/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 2.6.23-rc8+cfs (SMP w/2 CPU cores) Locale: LANG=en_US.UTF-8,

Hi list! I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche Telekom. Asterisk works well, but I have really often an high delay (I understand it since the other party speak some seconds before he hears my question and answer) and sometimes I hear an echo. I really don't know what can I check and what can be the problem. The problem exists since a ver...

Am 13.06.2015 um 13:54 schrieb Luca Bertoncello: > I think there are many german users in this ML, that use Asterisk with the > new line of Deutsche Telekom (Magenta Zuhause). I don't think so. Most users will use the router provided by Telekom. Anyway, after 15 seconds of Google'ing for Magenta Zuhause and SIP, maybe this will help you: http://wiki.innovaphone.com/index.php?title=Howto:Deutsche_Telekom_Magenta_Zuhause_SIP_Provider...

Zitat von Sebastian Kemper <sebastian_ml at gmx.net>: Hi Sebastian > Brian suggests to check the SIP traces. You can either enable SIP > debugging in Asterisk like so: > > sip set debug on > > Or you could run tcpdump and capture the SIP traffic. > > The first option is probably the easiest. I tried with sip set debug 42 sip set verbose 42 The result was

...ke with Telekom. They tried to "reset my DSL port" (whatever it means). As result I was without Internet and phone for over an hour... Then I tried to call my cousin in Italy and the call was NOT dropped after 15 minutes... I'll try this evening again. Maybe it was a problem by Deutsche Telekom... Thanks Luca Bertoncello (lucabert at lucabert.de)

Am 11.06.2019 um 20:42 schrieb Antony Stone: Hi Antony, > I think the main question here is: how are you connecting Asterisk to the > telephone system? Via VoIP... > You mention that you're on DSL from Deutsche Telekom, but is the call going > over this DSL link to soem SIP provider, who then connects you to the PSTN, or > are you connecting Asterisk locally to the phone line via some ATA device? Deutsche Telekom uses since years just VoIP. No ISDN, PSTN, and so on... :( I'm connecting to th...

Hi list! My Problem: all calls to international numbers will be dropped after exactly 15 minutes... I have a VoIP-account by Deutsche Telekom. This is what I see when I call someone (my parents) and the connection will be dropped: == Using SIP RTP CoS mark 5 -- Executing [+39015222222 at default:1] Set("SIP/00493511111111-00000125", "newNumber=0039015222222") in new stack -- Executing [+39015222222...

>> It doesn't really depend on your sip.conf and Asterisk. Your gateway/router >> will be the major problem. My summer project will be to look at session > Are you sure? > Right now I'm using an italian SIP-Provider (Messagenet), configured in my > sip.conf and I can receive calls without any problem... > So, I don't think, I have to expect problem on my NAT

Markus <universe at truemetal.org> schrieb: > I don't think so. Most users will use the router provided by Telekom. These users do NOT use Asterisk on theis Telekom-line... I asked for someone using Asterisk on Magenta Zuhause... :) > Anyway, after 15 seconds of Google'ing for Magenta Zuhause and SIP, > maybe this will help you: I already know these links, and I