similar to: Calling a script when new users are added or a user changes his password

Zitat von Remko Lodder <remko at freebsd.org>: Hi Remko, > Emails can only be read if they are authenticated / authorized in > someway to access 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

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...

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)

Hi again! I always try to get my mobile phone work with my Asterisk. I tried to install Asterisk on my PC (with public IP), but it has problems, too... I think, my UMTS-Provider doesn't want to connect to dynamic IP or my DSL-Provider does not want it, too, since I have no problem to connect and get a very good audio quality if I connect to other SIP-Provider or to an Asterisk (SAME

Zitat von Luca Bertoncello <lucabert at lucabert.de>: > I removed the double browseable, but the situation didn't change... > > But I have notice something stranger: the problem just happen with two users > in the "Domain Admins"-group. > With another user, not in this group, new created files and directories have > the right owner... Well, I noticed right

Hi again I'm trying to authenticate a user against an LDAP Server (well, our AD, but it can LDAP). This is my configuration: hosts = my.server.local auth_bind = yes ldap_version = 3 base = CN=Person,CN=Schema,CN=Configuration,DC=company,DC=local scope = subtree user_attrs = \ =home=/home/imapproxy/%u, \ =mail=maildir:/home/imapproxy/%u pass_attrs = uid=%u, userPassword=%w

Hi again! About my previous E-Mail... I though about it and I think, that maybe I'm just very stupid... Since I called an INTERNAL number, Asterisk tried to call it. I tried right now to call an EXTERNAL number (using my context [myproxy]) and the behavior is NOT the same... Not 100% correct, but it tries the right way... Now my problem is to check in my dialplan if the peer, that

Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option 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

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

Rowland penny <rpenny at samba.org> schrieb: > Hmm, not what I thought, but if that is the case, two more questions: > > Why are you using the depreciated ntvfs ? So I found in an HowTo (I don't have the address anymore...) > If you joined as a DC, why do you have this in smb.conf: server role = > MEMBER SERVER What have I to use? Thanks Luca Bertoncello (lucabert

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Monday 06 Jul 2015, Luca Bertoncello wrote: >> Well, but for voice quality, which codec is better? >> alaw or gsm? > > A-law is better for voice quality (sorry, thought my original > explanation was > obvious). But note that if the destination is a mobile phone, GSM will be > used anyway, at

Am 23.06.2020 08:43, schrieb Luca Bertoncello: And another thing, I discovered right now... > Could you suggest me something to restrict the problem? > Currently, I think the problem can be: > > 1) on Asterisk > 2) on my Gateway/Firewall A couple of years ago I added this entry in my firewall: /sbin/iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS

Rowland penny <rpenny at samba.org> schrieb: > Can you be a bit more specific, it sounds like you mean users can logon > somewhere even though Samba isn't running, or do you mean that windows > users are not running a logon script when they log in. No, of course Samba runs... I mean, Windows boots and starts a Startup-Script (or will shutdown and call a Shutdown-Script).

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 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

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) 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

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: == Problem setting up ssl connection: error:140760FC:lib(20):func(118):reason(252) [Jun 5 20:16:25] WARNING[20826]: tcptls.c:669

Hi, list! I use Dovecot 1.1.14 at work. It is configured to manage IMAP and IMAPs. Exim is the MTA of the Network. Of course, I installed SpamAssassin and ClamAV to check all incoming E-Mails. An, of course, some E-Mail marked as Spam are Ham and some E-Mail marked as Ham are Spam... To teach SpamAssassin I create for every user two folder: "DochSpam" (for Spam E-Mails not marked)

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

Hello, there is no need to change canreinvite for provider configuration. Do not change MTU. Probably there will be another problem. I expect packet size 1466 would pass and higher will have the same result. It would be interesting to make the same test from the outside towards your asterisk with size 2 bytes larger the highest you are able to ping. Marek 2020-06-22 22:26 GMT+02:00, Luca

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)