similar to: What do you do to keep asterisk alive?

> On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson <oej at edvina.net > wrote: >> >> 7 feb 2010 kl. 15.09 skrev Per Jessen: >> >>> Thomas Winter wrote: >>> >>>> Hi, >>>> >>>> my Asterisk on debian lenny died after 80 days. >>>> >>>> server kernel: [7572666.186852] asterisk[3673]:

Over the last two weeks, we have had at least two "incidents" where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further).

Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian

I'd like to be able to search the list archives when I'm reading someone's message to put what they say in context based on what they've said, and what others have said in conversation with them, in the past. It would help me figure out whether to trust some submitters on some issues, and just learn more from the community's collective/cumulative research and discussion. Is

I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a little dim if they believe they can openly go about borrowing email-addresses like this. /Per Jessen, Z?rich -- http://www.spamchek.com/ - managed email security.

from extensions.conf: exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) I basically try to lookup the CLIP and attach a name for each inbound call. This works fine, except when I have just restarted asterisk - at which time I've more than once seen the message from the subject. As far as I can tell, with my Set(CALLERID), I should always have an argument in the DB

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

I've got a Grandstream 487 in a home-office. The phone-side is working fine, but the user is complaining that his internet connection keeps disappearing. The Grandstream is set up as NAT router, and there's just one PC hanging off the LAN. Has anyone experienced anything similar? /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

About two weeks ago there was a thread about asterisk suddenly dying - I posted a response that the same happens to my asterisk about once a month, sometimes more. Someone suggested using 'safe_asterisk' (and get hold of a core dump) which sounds like a good idea, but one thing I can't figure is how to get "module reload app_queue" executed automatically at startup?

Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070809/ddaed76b/attachment.htm

I have a queue defined which I use like this: exten = _X.(reception),n,Ringing exten = _X.,n,Queue(enidan,t,,,20) exten = _X.,n,Voicemail(443,u) exten = _X.,n,Hangup() When I start asterisk, this queue doesn't work - -- Executing [4439000@Business:3] Queue("mISDN/3-u0", "enidan|t|||20") in new stack [May 14 13:53:59] WARNING[17860]: app_queue.c:3541 queue_exec: Unable

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I

I have the following dialplan: ; calls prefix by '8' are recorded exten = _8[01]./_251,1,Set(something=shortened) exten = _8[01]./_251,n,Set(WAV=filename) exten = _8[01]./_251,n,Monitor(wav,${WAV},mb) exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g) exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1} emailaddr) exten = _8[01]./_251,n,Hangup() The idea is that

This must have been asked before, but googling didn't help much. How do I define a callerid that contains non-USASCII characters? E.g. ?, ?, ?, ?, ?, ? etc. ? /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

Hi, I have just started to get dovecot list messages which I had not been receiving until today. How can I opt out (again)? regards, Michal Szymanski On Fri, Nov 30, 2018 at 10:42:22AM +0100, Per Jessen wrote: > Aki Tuomi wrote: > >On 30.11.2018 10.03, Per Jessen wrote: > > > >>Hi AKi > >> > >>I guess my address was re-subscribed then? I was subscribed

Michael Kliewe wrote: > Hi Per, > > On 25.08.2015 11:40, Per Jessen wrote: >> sieve and managesieve are both working fine, have been for years. Now >> I want to add a sieve_after script: >> >> /etc/dovecot/conf.d/90-sieve.conf : >> >> plugin { >> sieve = ~/.dovecot.sieve >> sieve_dir = ~/sieve >> sieve_before = >>

Hi, My Gigaset S450ip allows 2 simulatneous calls when each incoming call are targeted to different phones. When both calls target the same extension, the second one is forwarded to voicemail. I couldn't check yet SIP messages but has anyone met this limitation (one simultaneous call per phone) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone have a quick recomendation for a 2-port (ie. 4 channels) BRI (ISDN2e) card in the UK? My usual source has 1 port and 4 port, but no 2-port cards... Thanks! Gordon

Dear users, I think I may found a bug in the voicemail module of Asterisk 1.4.2! Outgoing email notifications should use a real existing domain (let's call it domain.real) instead of the local domain (domain.local) so that some mail servers won't reject the mails. That's why I've set the serveremail option in voicemail.conf to asterisk@domain.real. Unfortunately Asterisk is

Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was