similar to: Asterisk without D-Channel possible?

Hello everyone, My setup: Debian squeeze Asterisk 1.8, DAHDI, libpri, compiled from source TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800 I'm trying to get MWI for Voicemail working. In the same server I have also got an Eicon DIVA PRI card for testing purposes (it is integrated via CAPI and the chan-capi channel driver into my Asterisk). MWI works just fine there. I

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

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.

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

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 done that to improve security and to be able to assign all calls coming in

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

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

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

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

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

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

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

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

Am 13.06.2020 um 18:06 schrieb Michael Keuter: > So the call used Alaw as Codec. Yes, so seems it to be... It should has the better quality... But the calls done using my mobile phone in VoIP with the Asterisk have better quality as the calls done using the normal VoIP-telefon... I'm really puzzled... Luca Bertoncello (lucabert at lucabert.de)

Hi list! I'm using Asterisk 13.14.1 from Debian 9 repositories. The provider is Deutsche Telekom und Messagenet (just for receive). I can call and receive calls, but I have a little problem: there is a "delay" of about 1-1,5 seconds between the time the voice is sent and the time when the voice is received, so that it happens very often that the peer does not get my voice and try

-----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'm very new in Asterisk and VoIP, and of course I have a problem... :) Well, my problem is, that Deutsche Telekom wants me to change my ISDN to VoIP... :( I must do that, since I have no alternative. Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can configure my two numbers by Deutsche Telekom and I got now an extra number from Messagenet.it. Now the

On Monday 15 June 2020 at 18:55:23, Luca Bertoncello wrote: > Absolutly *no changes* on the behaviour compared with my Thomsons... Okay, I'm glad we can rule out the specific make / model of phone - that would have been bizarre. > I try to summarize: > > 1) Phones are not the problem, since 3 phones of 2 different > companies/model have the same issue. Good (if you see

-----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

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