similar to: Getting a list of availabe SIP-Header on phone

Patrick Laimbock <patrick 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...

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)

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > BLF is an interaction between the phones and the server. You need to > configure function buttons on the phones to display the presence state of > individual peers that have been set up on the server. > > This command in the asterisk cli will help you: > > core show hints > > If you see an entry for the peer then

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

Hi again! With the "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)

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

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

Steve Edwards <asterisk.org at sedwards.com> schrieb: > 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 > discussion looks relevant as well). > >

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 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:

On Sun, 31 May 2015, Luca Bertoncello wrote: > Now, it would be nice, if I can signaling on the phone which number will > be called, so that, for 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

2015-06-05 12:21 GMT-06:00 Luca Bertoncello <lucabert at lucabert.de>: > Hi list! > > I'm trying to configure my Asterisk to accept SIP-TLS connections, too. > > I followed this HowTo: > > http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ > > But as soon I try to connect to my Asterisk using SIP-TLS I get on > Asterisk-CLI: > >

2015-06-05 14:29 GMT-06:00 Luca Bertoncello <lucabert at lucabert.de>: > I think it is a problem on Asterisk for OpenWRT... :( > > Regards > Luca Bertoncello > (lucabert at lucabert.de) > compilation problems with the module srtp , check the module module show like srtp -- rickygm http://gnuforever.homelinux.com

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

Hi again! I'm thinking about using my mobile phone to receive (and send) calls when I'm not at home (for example in holiday). I can make my Asterisk reachable from Internet, of course, or I can use a VPN, that's not the problem... My question is: can I log in to the same account from more than one device? If yes, I can just configure my mobile phone with the same login of my

Zitat von Israel Gottlieb <isrlgb at gmail.com>: Shalom, Israel! > Using chan_sip you need to create another ?user aand then dial both > > Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I installed Asterisk... :( I'll create another user. Thanks Luca Bertoncello (lucabert at

ricky gutierrez <xserverlinux at gmail.com> schrieb: > Hi lucas , dou you try this: > > https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Tested right now. Same problem... I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucabert at lucabert.de)

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 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

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon