similar to: Asterisk and Deutsche Telekom

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

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:

Hi! I have a Asterisk installation to manage my phones at home (provider is Deutsche Telekom). It works, but very often the voice is "broken"... Yesterday during a call it was very difficult to understand what my partner sayd... It can NOT be a problem of other downloads/uploads, since in that moment there were no ones... I already had the problem in the past, solved it enabling the

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

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

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

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

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Karsten Wemheuer <kwem at gmx.de> schrieb: Hi Karsten! > the timeout value of 15 minutes directs me to an issue with session > timer. Try to refuse them by putting the line > session-timers = refuse > into the general context of sip.conf. Reload the sip stack with "sip > reload". Sorry, I forgot to mention that... I already have this setting:

"Brian ::" <bc at iptel.co> schrieb: > sip trace? Could you please explain? I'm not a VoIP-expert... Thanks Luca Bertoncello (lucabert at lucabert.de)

Zitat von Sebastian Kemper <sebastian_ml at gmx.net>: > I don't remember seeing anything looking like a SIP trace in your first > mail. Try > > sip set debug on > > instead of > > sip set debug 42 > > I don't think there's a sip debugging level like 42 in Asterisk. You can > either switch it on or off. Is it not this:

> 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

Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi again, > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the same > number 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.

Am 17.06.2020 14:37, schrieb Karsten Wemheuer: Hi Karsten! > The product is "All-IP" and not the SIP trunk, right? > The call starts normally and after about 15 minutes the quality is > disturbed? No, current we have Magenta Zuhause. Tomorrow we'll change to DeutschlandLAN IP (business contract). The quality is disturbed from the first second... I had the problem, that

Hi list! So, now I have a business contract and a technician was here to check the DSL... Nothing found, except that for 50Mbps I need now vectoring. Really nice... A couple of years ago I could get 50Mbps without vectoring. Of course, Deutsche Telekom said nothing about this change... Well, I got it working, and now I have 48Mbps down and 10Mbps up. I _REALLY CAN'T_ believe, that this is

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

Hi list! Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't connect to the remote Server (by Telekom) until today about 7:30. Well, it could happen... What I find really annoying was that I needed to restart Asterisk as I checked with sipsak that the Telekom-Server works... I think, this should not be normal... Can someone explain me why it happens and what I have to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > Don't use Port 5061, your SIP-port should be always even like 5060, > 5062, 5064 or 5066. Could you please explain why? I see in /etc/services, that 5060 is the port for SIP and 5061 for SIP-TLS, but I don't find anything for the other ports... Thanks Luca Bertoncello (lucabert

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 very long time, but in the