similar to: Extension record update for invalid uid=

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...

Zitat von Sec Adm <secadm2007 at gmail.com>: > In all cases you need an account I mean: if I just want to have info@ on the Server and all users accessing it via IMAP, I just need __ONE__ account. If I want to use a shared mailbox, __ALL USERS__ need an account on the server. Is it correct? Thanks Luca Bertoncello (lucabert at lucabert.de)

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > Looks like your phones do not 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

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)

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

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

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)

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)

Ishfaq Malik <ish at pack-net.co.uk> schrieb: Hi Ishfaq > Look into Busy Lamp Field/Presence > > Here's a starting point: > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my configuration: sip.conf: [general] allowsubscribe=yes subscribecontext = default

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

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)

> 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

Hi and happy new year! My question: - two extensions: 1111 and 2222 - an active call on 1111 - incoming calls to 1111 should be forwarded to 1111 (call advice!) and 2222 I know how can I forward an incoming call to more than an extension, but I have no idea how can I get the information, that 1111 has already an active call... I think, I need something like: exten =>

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 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, 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 =

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

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

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