similar to: IMAP disconnect before greeting / banner

> On May 29, 2017 at 9:27 PM Marcio Merlone <marcio.merlone at a1.ind.br> wrote: > > > Hi, > > I am running dovecot 2.2.22-1ubuntu2.4 on a ubuntu 16.04 server. It has > a valid Letsencrypt certificate but the problem also happens with a > self-digned one. > > Only openssl s_client -connect localhost:993 works fine and fast, while > all MUA's and

Hi, I am running dovecot 2.2.22-1ubuntu2.4 on a ubuntu 16.04 server. It has a valid Letsencrypt certificate but the problem also happens with a self-digned one. Only openssl s_client -connect localhost:993 works fine and fast, while all MUA's and telnet does not. Telnet timeouts waiting for banner after a minute or so: root at netuno:~# openssl s_client -connect localhost:993

Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz

Hello Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon <benoit.panizzon at imp.ch>) a écrit : > Hi List > > We have some CPE which run an embedded asterisk 13 with chan_sip. > > Unfortunately, when a registration is rejected, those stop trying. > > I am familiar with pjsip which allows to configure: > > auth_rejection_permanent=no > > How

On 2013-06-16 21:46, Timo Sirainen wrote: > On 14.6.2013, at 9.15, Benoit Panizzon <benoit.panizzon at imp.ch> wrote: > >> Is there a way to get quota-status to also use the proxy feature to >> request >> the quota information from the correct machine? > > Looks like this is a missing feature. I first thought quota-status > would go through doveadm

In article <200603181001.08589.benoit.panizzon@imp.ch>, benoit.panizzon@imp.ch says... > Thank you for the hint. Now finaly I can 100% reproduce the problem. Yes, if I > hang up during Playing 'conf-onlyperson' my machine freezes. It's not a GSM > Enconding problem as I suspected first, this happens with every encoding. > > magma*CLI> > -- Executing

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

So long as the tcp socket is open your SBC should send the call back over the same socket. Now it can be that your SBC is seeing the socket as timing out. If you are using Kamailio you can have it send tcp keep alives every so often so that the socket stays up. On Fri, Dec 27, 2019 at 10:41 AM Benoit Panizzon <benoit.panizzon at imp.ch> wrote: > Hi List > > I wonder how SIP via

Hello We would like to customize the voicemail menues. So the intro should not be played if some user has recorded an own greeting message and we would also like to remove some options from the menue. Is this all hardcoded or is it somehow possible to redefine the voice menues and the order how messages are played via voicemail.conf? Mit freundlichen Gr?ssen Benoit Panizzon -- I m p r o W

Hi Looking at the source of app_voicemail.c there are many statements like: ast_debug(1, "%s doesn't exist, doing what we can\n", prefile); Where do I have to enably this to be showed in the console or logged to a file by logger. core set debug does not seem to help here. Well, my actual problem is, that if a customer has recorded his own greeting, he

On Fri, Jan 10, 2020 at 12:25 PM Benoit Panizzon <benoit.panizzon at imp.ch> wrote: > Hi List > > I have been pondering over a problem to use an asterisk server behind > an SBC unable to successfully handle registrations. > > Now I observed something strange which I suspect might be a bug on the > asterisk side. > > The SBC originates is register from Port 6011 to

Hi After migrating from 1.2.4 to 1.2.5 I noticed that: show application dial does not show the 'R' option anymore. Has this become an undocumented feature or has it gone completely? Mit freundlichen Gr?ssen Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00

Timo, It looks like Dovecot 2.0 appends an APOP challenge to the POP3 greeting even if APOP is not an enabled auth mechanism. Is there any way to disable this? We don't support APOP, and the challenge includes the private hostname of the server, which we'd rather not have in the banner. It looks like get_apop_challenge in 1.2 returns NULL if APOP isn't supported, which causes

Hi all I have a very strange problem here... I use a hfc-s card with mISDN in NT mode with an ISDN Phone connected. When I make a call, the phone rings two or three times and then misdn runs into a timeout... I don't know where to set that timeout, but it's way to short for the called to pick up the phone. If the destination phone is picked up, then everything is allright and the

Hi All I'm tracing a very strange problem which I could reproduce with different versions up to 1.2.5 (sorry, didn't update to a newer one yet). Scenario 1: Problem does not occure. ============================= Sip Phone registered directly to the Asterisk. exten => i,1,Zapateller() exten => i,n,Playback(invalid,noanswer) exten => i,n,Hangup() Works like expected. I dial an

Hi all I have a problem that starts driving me crazy... Win2k3 ADS, added some attributes like loginshell, gid, uid etc. Unix clients use NSS_LDAP to get 'passwd' data and kerberos to authenticate users. Authentication does not happen via LDAP. winbindd is used to autocreate sid => uid/gid mappings. This worked very fine with samba 3.0.14a. Upgraded to samba 3.0.23 Now the owner

Hi all There seam to be a very short timeout waiting for digits being dialed. (about 6 seconds). Is there a way to increase that time? I have a phone with integrated address book and my fingers are just not fast enough to open the menue, select an entry and hit 'dial'. -Benoit-

Hi out there To play the correct announcement in app_voicemail I whould be able to read the SIP Diversion Reason which ist sent by another PBX: Invite contains: Diversion: <sip:+41315995003 at 157.161.10.190>;reason=no- answer;privacy=off;counter=1 Asterisk Logs: RDNIS for this call is is +41315995003 (reason no-answer;privacy=off;counter=1) From what I see in the source of chan_sip

Hi all I don't manage to get asterisk 1.2.5 or 1.2.6 running with the zaphfc driver.... The scripts from junghanns.net do download a very old libpri and asterisk version which is too buggy for me to use. Isn't there an acutal patch to get zaphfc support in *? -Benoit-

i want to be able to suppress the banner from the client side (ssh/slogin/scp/sftp) but i don't see a way to do it cleanly. for example, if there were a -B flag that suppressed the banner that would be alright. i did try -q, but that suppresses all stderr, which is unacceptable since i do want to see the error output when ssh fails to know why it failed. another idea would be to have -q