similar to: Dovecot as Proxy for __MORE__ Exchange-Servers

Hi List! I **HATE** Exchange-Server. I think it's not able to manage E-Mails, but we have to use it at work... Well, we need to read the E-Mails from outside, so I configured Dovecot with IMAPC to connect to the Exchange-Server. It works, but I have a problem... If I receive an E-Mail, and I read it from my phone (for example), I see in Outlook that the E-Mail as been read. If I move

Hi list! Finally I got the LDAP-Authentication work (it was a problem of the OU-Path... :( ). Now I can authenticate the user against the AD and forwarding the IMAP-Connection to the Exchange Server. Wow! My next problem: we have TWO ADs and TWO Exchange-Servers. The first AD has the users for the first Exchange, and the second AD for the second Exchange. I defined two files so:

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

> Am 13.06.2020 um 13:36 schrieb 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

Hello, if you need clampmss then it is highly probable there is a PMTU discovery problem. The clampmss does not work for UDP. 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. You

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

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...

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

The easiest solution may be to strip the leading zero's off your caller ID before your caller enters the Voicemail app to leave you a message. ExecIf(REGEX("^[0][0]." ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2})) On Fri, Jul 3, 2015 at 10:53 PM, Luca Bertoncello <lucabert at lucabert.de> wrote: > Hi list! > > Yesterday I set up a voicemail on my

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

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: Hi, > GSM is the native codec used for calls to mobile phones; it uses lossy > compression to achieve a low bit rate. > > A-law is the native codec used by physical exchanges on the land line network > (PSTN and ISDN). It is non-lossy. It works by arranging the "steps" closer > together near the zero

Hello, this is a correct response: >From 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,

If you allow anonymous search on AD maybe you can try to set auth_bind = no . a. On 22/06/15 17:19, Luca Bertoncello wrote: > 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 =

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

Hello, this could be ip address of the different interface on the same box. I think it works like expected. The only exception would be if the sip peer ignores the icmp packet unreachable. But I doubt this is the case. Anyway you get problems also when calling to LTE phone without using sip provider. Let first concentrate on these calls LTE to LAN. Are you sure you do not block incoming icmp

Hi Luca, It's not the A number you have to look at if you want to know how a call comes into the dialplan and then goes out again. You want do know in which context a call arrives. That depends on things like the IP address (peer), username/password (friend) or other things. I suggest to read up on that using the Internet (there are e.g. wiki articles about this subject) or a book (e.g.

Hi Luca Y need to check your wifes codec priority list -seems to be GSM on the first place. Luca Bertoncello <lucabert at lucabert.de> wrote: >Hi list! > >I noticed that when the phone of my wife calls the gsm codec will be used, >but if someone calls the phone, alaw will be used: > >00493511111111 calls 00493512222222: >OpenWrt*CLI> sip show channels >Peer

> Am 14.06.2020 um 16:38 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > 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

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Thursday 11 Jun 2015, Luca Bertoncello wrote: >> Now my problem is to check in my dialplan if the peer, that originate >> the call, is reachable, and if not, to give an error... >> >> Is there any function to know if the peer is reachable? > > The peer that *originated* the call *must* be

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