search for: st2022

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 problems: 1) It seems that I can't configure my ST2022 to have two profiles and both are running on different servers 2) I want that when a number will be called, bo...

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)

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > Do you have a link to the user guide for your exact phone model? Unfortunately not... I have a Thomson ST2022, but I can just find in Internet manual for the ST2030... Regards Luca Bertoncello (lucabert at lucabert.de)

...k cli will help you: > > core show hints > > If you see an entry for the peer then the server is set up correctly and if > the Watchers column > 0 then you have set up the phone correctly. Unfortunately the Watchers are 0... And I didn't find any option on my phone (Thomson ST2022) to enable the BLF... Any other idea? I wrote a little expect-Script to send the phone an advice and having an LED blinking, but I think it is a little bit exaggerated... Thanks Luca Bertoncello (lucabert at lucabert.de)

> 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 problems: > 1) It seems that I can't configure my ST2022 to have two profiles and > both are running on different servers > 2) I want that...

...sed): > > 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, > to your Asterisk server, over your home *wireless network*, to place a call to > some external number, you have a conversation and *the quality is excellent*. > > 2. You use your *Thomson ST2022*, which is also registered by SIP, to your > home Asterisk server, over your home *cabled* network, to place a call to some > (the same???) external number, you have a conversation and the quality is *not > excellent*. > > > Is that an accurate summary of your situation? Not...

...ck at laimbock.com> schrieb: > On 12/30/15 12:24, Luca Bertoncello wrote: > > Ishfaq Malik <ish at pack-net.co.uk> schrieb: > > > >> Do you have a link to the user guide for your exact phone model? > > > > Unfortunately not... > > I have a Thomson ST2022, but I can just find in Internet manual for the > > ST2030... > > The administrator manual can be found at: > http://www.manualslib.com/manual/909341/Thomson-St2020.html?page=5 > > To download click the green Download button at the top. Hi, Patrick! Thank you very much! Unf...

...: UNKNOWN Useragent : Reg. Contact : Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Refuse Sess-Refresh : uac Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No VoIP-phone (Thomson ST2022): bpi*CLI> sip show peer 0049351xxxxxxx * Name : 0049351xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon Record Off feature : automon Subscr...

...;call pickup"-function I can now pickup a call directed to another phone in my Asterisk. Very nice. My problem, now, is that I can't see on my phone, that the other phone (in another room) rings. Is it possible to signal the incoming call on other extension? I use two phones "Thomson ST2022". Thanks a lot Luca Bertoncello (lucabert at lucabert.de)

...example, if I receive a call for +493511111111 I get a message on the display or the phone ring with a particular tone, and if I receive a call for +493512222222 the phone write something other on the display or ring with another tone. Is it possible? Maybe it depends from phone... I use a Thomson ST2022. Thanks a lot Luca Bertoncello (lucabert at lucabert.de)

...> Qualify Freq : 60000 ms > Keepalive : 0 ms > Sess-Timers : Refuse > Sess-Refresh : uac > Sess-Expires : 1800 secs > Min-Sess : 90 secs > RTP Engine : asterisk > Parkinglot : > Use Reason : No > Encryption : No > > VoIP-phone (Thomson ST2022): > bpi*CLI> sip show peer 0049351xxxxxxx > > > > > * Name : 0049351xxxxxxx > > > Description : > > > Secret : <Set> > > > MD5Secret : <Not set> > > > Remote Secret: <Not set> > > &gt...

Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca

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

.../0.190 ms, pipe 2 But now I made a test with a friend of mine, and I think the results are very interesting... So, we configured his mobile phone (Android) to use my Asterisk as peer. We created also a VoIP account on the phone. The phone was *NOT* in my WLAN. The friend called my phone (Thomson ST2022 in local LAN). This was a VoIP call inside Asterisk (two peers speaking together). Deutsche Telekom was *NOT* used now! I can hear very good the friend, without "broken voice", but *he* just hear "like a robot with sore throat" and can't understand any word... The same if I...

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

...l for +493511111111 I > get a message on the display or the phone ring with a particular tone, > and if I receive a call for +493512222222 the phone write something > other on the display or ring with another tone. > > Is it possible? Maybe it depends from phone... I use a Thomson ST2022. You can fiddle with the caller ID to change what is displayed on the phone. You can fiddle with the ring tone by phone specific configuration and phone specific SIP headers (sipaddheader(Alert-Info: ...)). These seem relevant: http://www.voip-info.org/wiki/view/RTTTL+melodies+for+ST2030 (the...

Hi list! I read the pages that Steve sent to the list. It sounds nice, but I didn't found any documentation about available SIP-Header on my phone (ST2022, not ST2030!). Is there a possibility to ask the phone which header it understand? Or to get this list in other ways? Thanks a lot Luca Bertoncello (lucabert at lucabert.de)

...nderstand (differences emphasised): 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, to your Asterisk server, over your home *wireless network*, to place a call to some external number, you have a conversation and *the quality is excellent*. 2. You use your *Thomson ST2022*, which is also registered by SIP, to your home Asterisk server, over your home *cabled* network, to place a call to some (the same???) external number, you have a conversation and the quality is *not excellent*. Is that an accurate summary of your situation? Antony. -- Just when you think...

..., Luca Bertoncello wrote: > 1) I have an Android phone, using the integrated Android VoIP-subsystem, > connected to my Asterisk at home, over LTE or other network *outside my > home network*. > I called my mother using this method... The quality was excellent > 2) I have a Thomson ST2022 connected to my Asterisk over Ethernet > (cabled network). If I call for example my mother or my parents in law, > the conversation is "broken", eg: both partner can hear little > "interruption", about 1/10 seconds in the conversation... I would like to see a much simp...

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > The hints have to be in the same contexts in extensions.conf as defines in > the sip.conf subscribecontext which can be set per peer. Well, [anika_incoming] will be included in [default], of course... But I tried to define anika_incoming in subscribecontext, too. No changes... > Also, have you configured the phones as well? What do