similar to: Asterisk spontaneous reboot

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

Hello thank you for your answer. If I read your (and others) reaction correctly I can conclude that this is an Asterisk problem and not a problem of MySQL or dialplan logic ? You should know that the MySQL database is heavily questioned : mysql> show status like '%onn%'; +--------------------------+--------+ | Variable_name            | Value  |

Hello using Asterisk 1.8.32. I notice that there is a spontaneous reboot of the Asterisk system from time to time. When I look in the logs (verbose file) I noticed that every time this occurs it's at a moment that there is a MySQL action, be it a lookup or an insert/update/delete. I must say I do have some MySQL queries that occur in my dialplan when a call comes in, to look up

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 11-08-16 18:03, Matt Fredrickson wrote: > On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote: >> My main reason not to upgrade to Ast 13 is because I'm afraid of losing >> functionality as there are certain functions deprecated/replaced. This can >> also cause headache :-) >> >> I will do so if there is no other option.

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, ????? ?????? wrote: > > Try delete nat from 770000wrtc settings ice should do the same > > > On Aug 11, 2016 10:00 PM, "Jonas

Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init

On 10-09-16 00:50, Richard Mudgett wrote: > > > On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when I type on the Asterisk CLi 'queue show', I first get a list > of my queues and then the following : > > > failed to extend from 240 to 327

Hello, is there anyone who can point me to correct information ? Following http://pbxinaflash.com/forum/showthread.php?t=9042 and http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql > Extending CDR does not result in a working environment for me. Any feedback appreciated. Kind regards, Jonas. -------- Original Message -------- Subject: [asterisk-users] Asterisk 1.6.2.10 & CDR

On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >

Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120202/7954fe9e/attachment.htm>

My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-) I will do so if there is no other option. But still, I don't see why Ast 13 would differ so much in this case ? If ICE and NAT is working (not causing problems) why should Ast 13 bring me audio and Ast 12 don't

Is there any way to do this? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120307/77764e4b/attachment.htm>

Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number. I expect here that colleague B would see the external calling number on the screen

Hello, are there any known reasons why Asterisk would disconnect random calls ? My server uses 1,5 GB out of 8 GB RAM My server uses up to 35% CPU at peak There are about 40 concurrent calls. I have 300 RTP-ports available. I just see the call ending, as if one of the connected parties hung up but that is not the case ! So what could be a bottleneck ? Any known reasons for random hangup ?

Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip