search for: lucabert

...terisk (SAME configuration!!) installed on my Server... Well, I will try to configure the Asterisk on my Server to act as "proxy" so that all phones at home talk with my Asterisk at home (now called "wrt", and my mobile phone talk with my Asterisk on my server (now called "lucabert"). I followed this HowTo http://sysmagazine.com/posts/125303/ and I got both Servers talking together. I can call my mobile phone (logged in at "lucabert") from a phone logged in on "wrt" and a phone at "wrt" from my mobile phone at "lucabert". Wonderf...

...is problem? > I probably counted the size incorrectly. So you are able to ping with > size 1464 and not with 1466. How about trying same ping sizes from the > internet towards your site? I mean trying to ping from sites with > higher MTU than yours without lower MTU links in the path. lucabert at ns:~$ ping -4 -M do -s 1465 bpi.d.lucabert.com PING bpi.d.lucabert.com (93.241.91.232) 1465(1493) bytes of data. From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set (mtu = 1492) ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ping:...

...ss the store. That means you might need to share the > info@ credentials with the other > people so that they can read it over imap or webmail etc. That is self-evident and it is not a problem. I can't understand what you mean... Do you see a problem in that? Thanks Luca Bertoncello (lucabert at lucabert.de)

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

...56.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set > (mtu = 1492) > > So PMTU discovery is working. No problem here. You got correct message > to lower the packet size from 62.156.246.57. This is probably the last > hop before your site. No, the last hop is 62.156.246.65: lucabert at ns:~$ mtr -4nr bpi.d.lucabert.com Start: Tue Jun 23 10:10:16 2020 HOST: ns.lucabert.de Loss% Snt Last Avg Best Wrst StDev 1.|-- 185.242.112.1 0.0% 10 0.4 1.1 0.3 4.4 1.2 2.|-- 84.200.230.82 0.0% 10 0.8 0.7 0.5 0.8...

...in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, maybe with an example? Thanks Luca Bertoncello (lucabert at lucabert.de)

Hi again! With Dovecot 2.2.9 authenticating against the Active Directory I have following problem: - if I login using LOGIN, PLAIN or CRAM, the username is REALM\login (in my case: CCH\lucabert) - if I login using GSSAPI, the username is just login (in my case: lucabert) this makes the access to the mailbox very difficult, since I don't what can I write in mail_location... If I login with LOGIN, I can write: maildir:/home/mailboxes/%4.u/Maildir but of course it does not work if I...

...22222&SIP/00493511111111,,R) of course it works... Now the problem is, that when a phone get the call, on the other phone I get "1 missed call"... Is it possible to avoid that and signaling the other phone, that the call was not "missed"? Thanks a lot Luca Bertoncello (lucabert at lucabert.de)

.... If it rings and is not answered it counts that as a miss. The > only way to avoid it is to not ring it. So instead of simultaneous ringing > you can do sequential. > > -- > Mehmet Avcioglu > mehmet at activecom.net > > > On Jun 4, 2015, at 11:21 PM, Luca Bertoncello <lucabert at lucabert.de> > wrote: > > > > Hi list! > > > > I configured Asterisk to forward the incoming call for a number to both > phones. > > I wrote that: > > > > exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,R) > >...

...y high port for SIP> --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128 and change the previous entry like: iptables -A FORWARD -p tcp -i intlan0 --tcp-flags SYN,RST SYN -j TCPMSS --clamp-mss-to-pmtu to limit the behaviour on the internal LAN... Your opinion? Thanks a lot! Luca Bertoncello (lucabert at lucabert.de)

...read the E-Mails via IMAP. Now the question: if I have 5-6 people accessing the account via IMAP, has Dovecot (2.2.13 from Debian repositories) problems? Is there other solution, if the users don't want to have an account on the server? Thanks a lot for your suggestion Luca Bertoncello (lucabert at lucabert.de)

...risk using SIP-TLS I get on Asterisk-CLI: == Problem setting up ssl connection: error:140760FC:lib(20):func(118):reason(252) [Jun 5 20:16:25] WARNING[20826]: tcptls.c:669 handle_tcptls_connection: FILE * open failed! And of course it does NOT connect... Any idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

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)

...n.Surendran at now-health.com> schrieb: > Have you tried NAT=force_rport ? OK, tried... I can transmit from my phone (aka: I hear my voice on another phone), but I'm not able to receive data (aka: I cannot hear what I say on the other phone). Other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)

...in Internet. But maybe my Provider does a NAT, too... Very strange is, that I have a very poorly audio-quality, if I use my cellphone in my WLAN and connect to my Asterisk. With THE SAME USER, but from a PC in the same Network, the audio quality is perfect. Any idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

...sip stack with "sip > reload". Sorry, I forgot to mention that... I already have this setting: session-refresher=uac session-timers=refuse > (I assume You are using chan_sip. I don't know how to disable session > timer in pj sip). I use chan_sip. Thanks Luca Bertoncello (lucabert at lucabert.de)

...t support it. And it is a very common feature. I think so... Maybe I can write a little program running on my PC to receive a message from Asterisk if someone calls the other phone... I'll think about that... Or maybe is there already such a program running on Linux? Regards Luca Bertoncello (lucabert at lucabert.de)

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > Look up fop2 Thank you very much, but I prefer a standalone application, if it's possibile... Any other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)

...62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set (mtu = 1492) So PMTU discovery is working. No problem here. You got correct message to lower the packet size from 62.156.246.57. This is probably the last hop before your site. Marek 2020-06-23 9:40 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>: > Am 23.06.2020 09:28, schrieb Marek Greško: > > Hi > >> if you need clampmss then it is highly probable there is a PMTU >> discovery problem. The clampmss does not work for UDP. > > Is there a way to check if I have this problem? > >> I pr...

Zitat von Aki Tuomi <aki.tuomi at dovecot.fi>: Hello Aki, > Or you can use shared mailboxes... > https://wiki.dovecot.org/SharedMailboxes/Shared Understand I right, that in this case, I __NEED__ all users to have an account on the server? Thanks Luca Bertoncello (lucabert at lucabert.de)