search for: definatley

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

...e the instructions that have been known to work; > Please read line by line and setup that way. > > http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup > > Dan > > On Sun, 6 Mar 2005, skamp wrote: > >> sorry still doesnt help with incoming calls, there is definatley >> something more wrong, my config was working fine until today and its >> worked fine for months. They have broken something. >> >> On Sun, 2005-03-06 at 02:23 -0600, Bartosz Wegrzyn - asterisk wrote: >> >>> [broadvoice-incoming] >>> type=peer >&g...

Life times of one brand of light bulb were observed. Life hours of nine light bulbs were 898, 720, 354, 405, 620, 54, 229, 306, 890. There were three more light bulbs that were still burning after 1000 hours. It is common to assume the life time of a light bulb follows an exponential distribution. With the observed data find the MLE of the parameter of the exponential distribution with EM

Forgive the term global if that''s not applicable in rails (i come from a php background) I have a browse controller with a couple of different methods def method1 @properties = Property.find_all #other stuff for that method end def method2 @properties = Property.find_all #other stuff for that method end def method3 @properties = Property.find_all #other stuff for that method end I

Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware

Hey guys and gals, I have a snippet of code: Net::SMTP(@host, @port, @from_domain) do |smtp| @emails.each do |email| begin smtp.send_message email.encoded, email.from, email.destinations @emails_sent += 1 rescue Exception => e # blah end end end What I want to do is: Say there are 4 emails. First email is sent OK On the second email smtp raises a IOError

I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last

When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi -> Asterisk --> network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this