similar to: Unusual message

Hi Harsh, sorry, but you're problem is not related to my patches. This may need further investigation. metze Am 31.01.2018 um 12:45 schrieb Harsh Kukreja: > Hi Stefan > > I am also one of the Sernet customer. Can you guide me how to run the patch > to fix the bug. > > I am running 2 DC's Sernet Samba 4.7.4 with 2 RODC's running Sernet Samba > 4.7.4. Whenever

Hi Stefan I am also one of the Sernet customer. Can you guide me how to run the patch to fix the bug. I am running 2 DC's Sernet Samba 4.7.4 with 2 RODC's running Sernet Samba 4.7.4. Whenever I run samba-tool drs replicate --fix --yes command on the DC it shows the below errors which cannot be fixed: Failed to remove deleted DN attribute fromServer : (65, "objectclass_attrs: at

Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110309/fe9d7bc7/attachment.htm>

Hi, Odd problem have just noticed in that when I call into the PBX DAHDI detects the call and hands it off to the extension, if I then hang up it still continues to process through the dialplan. This is what I have in chan_dahdi.conf: [channels] language=en echocancel=yes usecallerid=yes cidsignalling=v23 sendcalleridafter = 2 hanguponpolarityswitch=yes rxgain=2.0 txgain=3.0 progzone=uk

Hi, This is a real strange one and trying to phantom it out. One of our clients is connected to our Asterisk installation, from two sites, via VPN which works great. Every so often one of the sites VPN tunnel goes does for say a couple of seconds. When that happens all the extensions, including both sites, ring which is bizarre. Has anybody seen this before ? I only see two places in the dial

Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial

Hi I am running samba-tool dbcheck --cross-ncs --fix --yes command on my DC running 4.7.4 to remove the Deleted and Stale records which is failing Target GUID points at deleted DN '<GUID=3484ea7a-da88-4269-9b32-55a9c7faabf6>;CN=NTDS

Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. -- Thanks, Phil

Hello all, After numerous issues with Snom phones (360/370/870) potentially looking to migrate too Yealink as their product range looks very promising indeed. Are any of you using them with Asterisk ? How do they perform ? Do you use mass deployment at all ? Would be very interested to hear from you. -- Thanks, Phil -------------- next part -------------- An HTML attachment was

Hi, I have recently upgraded from CentOS 5.1 too 5.2 and now run Xen 3.0.3. What would be the best way to upgrade too Xen 3.2.1 ? I presume I would also need to change my network settings for xenbr0 aswell ? Any help would be greatfully appreciated. Regards, -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint: F57A 0CBD DD19 79E9

I have this running on my Asterisk server, and have opened up ports UDP/5060 and UDP/10000-20000 but for some reason when I try and connect too my SIP extension it does not work. Are these the correct ports ? -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver:

Hi, Now that I have Asterisk up and running I would like to find a good SIP gateway in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I am looking for peoples recommendations. Apologies if this is the incorrect forum for this type of request. Regards, -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint:

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Experiencing a problem when users attempt to forward a voicemail from within VoiceMailMain(Option 8) I see the console message: Couldn't not find mailbox XXX in context default As why are running in a multi-tenant environment voicemail.conf has been separated into individual contexts. The users retrieve their email by dialing an extension which calls VoiceMailMail(XXX at VMContext) so how

Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be loaded. Yet I do have iksemel installed: ls -l /usr/local/lib/libik* -rw-r--r-- 1

All- Recently an Asterisk server we host was hacked and used to route some unauthorized calls. We have since improved our security measures, including installation of fail2ban. The interesting thing is the way in which this was discovered. The unauthorized calls were occurring intermittently last Thurs evening thru Sat morning. On Sat morning, some of our employees were attempting to log-in

I finally managed to get AxCrypt working consistently under Wine but I wondered if anyone out there could throw some light on the remaining problems, or suggest a workaround... I copied the files from a windows installed copy, just the executable directory containing AxCrypt.dll & exe,AxCryptM.dll,AxDecrypt.exe, Config.xml, notify.exe,Program.ico and Sigs.xml, into a directory called

Hi, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my extensions.conf I have a simple plan for testing :- [inbound-sip] exten => uxbod,1,Dial(sip/1001,20,t) exten => uxbod,n,PlayBack(uxbod) exten => uxbod,n,VoiceMail(1001@voicemail,s) exten => uxbod,n,Hangup() exten

Hi, Over the weekend tried to setup a test using the new app_calendar code but receiving the following error: [Jan 17 09:23:35] WARNING[27663]: res_calendar_icalendar.c:146 fetch_icalendar: Unable to retrieve iCalendar 'testcal' from 'https://office.test.net/home/teamshare at test.net/Calendar/': Server certificate verification failed: issuer is not trusted The target server is

Hi, Running Asterisk 1.6.2.11 and wondering what would be the best way to send an email when a missed call has occurred ? I believe you can modify [stdexten] is this still the case in V1.6 ? -- Thanks, Phil