similar to: No more room in scheduler

Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples:

On 7/5/19 10:50 AM, Doug Lytle wrote: > On 7/4/19 6:40 PM, hw wrote: >> This has again, and for no reason, ceased to work again after >> restarting asterisk.  No matter what I try, I can't create a >> certificate asterisk >> would verify. > > Have you considered using LetsEncrypt for a valid certificate? > > Doug > > What would be the point

I have a magical proposal for the anti-magic web framework. Controller methods do not use arguments and yet expect arguments. This is handled through this params hash because we don''t know in advance what parameters a client could pass to Merb. But in almost every instance, it is too much to know all the query parameters - one doesn''t care if the user threw in an extraneous

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

On Fri, 2006-10-13 at 12:52:38 -0400, Jon Weisman <jweisman@ibell.net> wrote: > Hi All, Anyone know how to generate random numbers in the > dial plan? I've tried using the RAND function but it doesnt > work. Basically I need to generate a random 5 digit number > everytime a particular extension is dialed and then save that > into

https://bugzilla.netfilter.org/show_bug.cgi?id=1328 Bug ID: 1328 Summary: Please allow ipset add and del via the /proc/net/xt_ipset mechanism Product: ipset Version: unspecified Hardware: x86_64 OS: All Status: NEW Severity: enhancement Priority: P5 Component:

Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil

Hi all, I''m wondering about my sanity here. There''s probably something simple I overlooked. First off, I have close to zero experience with Merb. 1. I configured use_orm :activerecord in config/dependencies.rb; 2. I have a file named app/models/measurement.rb which looks like this: class Measurement < Observation end 3. I have another file named

Hello! Oh, wise ones, ponder with me over two of the surprises that populate the universe! I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time. Here is all the config relevant to that phone: [murftest12] type=aor qualify_frequency=1992

On Sat, 2006-10-14 at 12:00 -0700, asterisk-users-request@lists.digium.com wrote: > Steve, > > Is RAND available in the latest trunk or do I need the 1.4 > beta? > > If I do show function RAND it says its not available. > > Thanks, > Jon Jon-- Forgive me, you didn't say which version you

On Wed, Oct 29, 2014 at 7:10 PM, sean darcy <seandarcy2 at gmail.com> wrote: > On 10/29/2014 08:06 PM, Matthew Jordan wrote: > >> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >>> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? >>> >>> >> codec_silk for Asterisk 12 will most

Hey, just posting in case this is of interest. I just recently wrote a Rails plugin to convert Rails Test::Unit cases to RSpec. http://www.davidjrice.co.uk/articles/2007/8/12/ruby-on-rails-plugin- test-unit-to-rspec-converter it''s not going to work 100% in every case, but it''s really helped me convert old tests to the lovely RSpec :) Best, Dave --- David Rice

Is test2spec still around? If so, are there any special instructions for running it against edge? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/rspec-users/attachments/20070504/bb164e29/attachment.html

Hi all, I'm having a problem with an asterisk server being unable to call certain cellphones and answering machines. Anytime the person answers the phone call, everything works well. But when the call goes to voicemail or an answering machine, I get the error message below: ===================================== *CLI> -- Attempting call on IAX2/u92613106/15551234567 for

Hi, Asterisk does not post CDR when dial status is CHANUNAVAIL. Can someone tell me what are the conditions under which CDR is not posted? Thanks Jim

Is there any code that I can cut/paste that will allow me to receive an SMS text on Asterisk? and, where can I capture the incoming text?

Hi, I suddenly started to receive an error like [Aug 4 14:26:57] ERROR[2477]: chan_dahdi.c:10515 dahdi_pri_error: No more room in scheduler [Aug 4 14:26:57] ERROR[2477]: chan_dahdi.c:10515 dahdi_pri_error: Asked to delete sched id -1??? and it went on till i reboot asterisk and dahdi services.. I wonder what caused this error, because there were no physical problem between pbx and asterisk

Hi, I noticed that Dahdi starting producing these error messages: ERROR[29250] chan_dahdi.c: No more room in scheduler ERROR[29250] chan_dahdi.c: Asked to delete sched id -1??? during which time I could not send any calls or receive calls on at least one of my Dahdi spans. The only way to clear the problem seemed to be to restart Asterisk. It appears to start after the following message

Hi there I just started my first Merb app and I''m running into some weird things in Specs. When I do: describe "create action" do it "should create a new project" do project_params = {:name => ''bekkunin''} Project.should_receive(:create).with(project_params) controller, action = post(''/projects'', :project

Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683&nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev