similar to: Asterisk as "Proxy" and more device for a number

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 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)

Am 13.06.2020 um 22:09 schrieb Antony Stone: Hi Antony > You are *assuming* that it's the codec causing the difference. Well, I really don't know what I can think, now... > We don't know that. > > Let me get this clear, to make sure I understand (differences emphasised): > > 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, > to

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 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

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: Hi Kevin, > If you want to go the Asterisk from scratch route, you would do well to > pick up a book on the subject. Since you seem comfortable with English, > "Asterisk: The Definitive Guide" is a good place to start. This will teach > you how to build an Asterisk system from the ground up. Depending on

-----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

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

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

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

Hi list! Another day, another problem... I'm checking with Nagios my Asterisk at home, and since yesterday I noticed that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours, so that I have a new IP every day), the peer of an VoIP-provider I use is UNREACHABLE. Yesterday I though it was a problem on the line, but today is the same, so I think it is something other...

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

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! 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

> 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

Am 22.06.2020 um 17:41 schrieb Marek Greško: Hi > try pinging your sip peer ip address following way: > > ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress} > > Post several lines and the statistics. root at bpi:/etc/asterisk# ping -n -M do -s 1300 -i 0.1 -c 100 tel.t-online.de PING tel.t-online.de (217.0.128.133) 1300(1328) bytes of data. 1308 bytes from 217.0.128.133:

Am 22.06.2020 um 17:01 schrieb Telium Technical Support: > I don't know if there was a prior email with more details, but.... > > Latency is as important as speed. Have you checked latency between your device and pop? What about QoS at your location, and does your ITSP support/respect QoS? That's a very good idea... Could you suggest me how can I check it? The Gateway is a

Hi all! Currently I'm using Messagenet, a SIP-Provider in Italy, to have an italian number via VoIP, _to receive calls only_. I use it to allow my friends and parents in Italy to call me in Germany without paying too much. This service was free of charge in the last years. Now will Messagenet beginning from end of november, to cancel this free service and only offer paying services (for

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

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > What kind of phone are we talking about, both yours that works and your > wife's that does not? Right! > Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM > Many phones will have a network test function built in to them to help you > determine if the phone