similar to: No subject

Hi all I'm trying to configure my Asterisk setup to load the musiconhold.conf file from an ODBC connection to MySQL, working through the example given in the excellent book "Asterisk: The Definite Guide". I'm using Asterisk 1.4.19 and MySQL 5.1.58. I've configured the ODBC bit and in my GeneralLog on MySQL I can see the asterisk user connecting and sending a few SQL

Hi all We have an Asterisk server that?s been running for a few years now without problems. We have IPTables running, as well as fail2ban and have followed all the security recommendations we have found. Every few weeks we get an attack that lasts about a minute or two, resulting in our AGI script being overloaded. What happens is that somebody seems to be trying to connect from our

In my assets I have intranet |_1.css.scss |_2.css.scss internet |_1.css.csss application.css application_internet.css application_intranet.css 1.css.scss 2.css.scss 3.css.scss intranet and internet are directories. In my layout I want to exclude the files under internet and intranet. I have created application_internet.css /* * application-internet.css * *= require_self *=

Hello all Over the last couple of months we?ve been experiencing a strange problem, which I?ve been unable to solve. We have an Asterisk 1.4.19 that?s been running happily for the last several years. All calls go through an AGI server from dialplan. On average we have appr. 3000 incoming calls/day. All calls go in via the AGI server, various sound files and menus are played, and every

I have just installed Rails 3.2.1 on my Xubuntu using the gem system and I try to make the tutorial http://guides.rubyonrails.org/getting_started.html and on section 4.3 Setting the Application Home Page I get the following error ExecJS::RuntimeError in Home#index Showing /home/xonx/rails/blog/app/views/layouts/application.html.erb where line #6 raised: (in

Hello all, Can anyone point me to any solid documentation on how to set up a production server using RVM, Passenger, and Apache2? I have been at this for a week with absolutely no luck at all so I thought I would post here to see if anyone has managed to get this to work and written down the steps they used. Right now with my current implementation, I am getting a 404 error when I attempt to

The latest version of Fedena - the opensource school management system is now available to public for free download. Official Blog post - http://projectfedena.org/blog/17-fedena-2-3-released Github - https://github.com/projectfedena/fedena Download instructions - http://projectfedena.org/download Fedena now powers more than 40,000 institutions around the world. It includes the most notable

Hi all, I'm looking at options for installing/writing PBX software and I came across www.vicidial.org<http://www.vicidial.org> which seems to do almost all I need - and is open source and all. I'd very much like to hear from anyone having experience with VICIDIAL, e.g. using it with different versions of Asterisk (the documentation only mentions * up to 1.6) Best regards Binni

Hello. I''ve installed cancan and bigbluebutton_rails, https://github.com/mconf/bigbluebutton_rails. The gem bigbluebutton_rails has some models and controllers, for example has bigbluebutton/servers_controller.rb. This controller has some actions like create, join, authorize, etc. I want that only users with role admin can create. What can I do? I need to put load_resource in

Having some trouble with sip behind a nat. So tried: stunaddr = numb.viagenie.ca in sip.conf. Didn't help so tried stun debug: asterisk*CLI> stun set debug on STUN Debugging Enabled STUN Packet, msg Binding Response (0101), length: 36 Found STUN Attribute Mapped Address (0001), length 8 Ignoring STUN attribute Mapped Address (0001), length 8 Found STUN Attribute Changed Address (0005),

I am trying to "dial" a GTalk, ie @gmail.com, address. I inscribed this address in jabber.conf on the buddy= line. Upon executing the Dial application, I hear only a brief brief ring, then nothing. What might be the trouble? As the Dial application starts trying, the JABBER chatter on the console includes some "INCOMING" entries that name IP addresses. The one with

I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]:

Thanks Joshua for replying ! What would you advise : - leaving STUN address empty, in rtp.conf, as "STUN is not required for ICE" - configure it with one public STUN (I'm using stun.voip.ovh.net for this but I don't know how this server really works) Cheers Le mar. 27 oct. 2020 à 09:53, Joshua C. Colp <jcolp at sangoma.com> a écrit : > On Tue, Oct 27, 2020 at 5:35 AM

Hi Ray, I was wondering if the "qualify" option is used [in sip.conf] to keep a connection (from the SIP phone inside the firewall to the Asterisk server outside the firewall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started "talking" first)? I'm not sure about that - I'm being

After struggling with one way audio issues as a result of STUN binding errors on both the Asterisk side and the Chrome side, we've decided to just simply go with a TURN relay for RTP packets until the issues are resolved. I configured rtp.conf so that all of the STUN related entries are commented out, and I use the following TURN configuration instead: turnaddr=numb.viagenie.ca:3478 ; ;

Hi Everyone. I'm in a quandry & don't know which way to go. - Obviously I'm an Asterisk newbie although I've been watching this list for over 2 years now. I've got an Asterisk box (actually, it's an AsteriskNOW box) up and running here at home. - It's on my home LAN - NAT'ed behind my LinkSys router. - On the same LAN I've got a Cisco 7940, 7960, and

Hello, I have a problem with a Grandstream being behind a symmetric nat. The box which does the nat is a german "Fritz Box". This one does nat for the internal network. In the internal network is a Granstream BudgeTone 100. The nat router has a dial-up connection, so ip changes on every dial-in. |------------| |------------| |--------| |Grandstream

Dear all, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr

Hi all, I am running into a weird issue where my application is giving me a stack level too deep error. I can''t seem to find where my code is throwing this error. On the machine I developed this code on, this error does not show up when I run rails s but when I move the code to another system to run it this error shows up. The system that this code runs fine on is a mac os x machine

I have two phones, one does not need stun, the other one needs. All settings are identically, except the number/password and said above stun - not stun I use codec in the order: g729 g711u g711a Any ideas, why the user can hear me, but I cannot hear him (stun) while the other user without stun has no problem. bye Ronald