search for: dsalama

...the correct value defined in AccountsController. However, the @title defined in the components template renders the value defined in ApplicationController. Isnt'' everything supposed to be rendered in the context of the controller/action requested? rhalff commented on the IRC: rhalff: dsalama, but a component is a controller itself, so it uses that context. your component probably extends application controller and you didn''t set a title in the component controller itself. To which I replied: dsalama: rhalff: you just hit the nail in the head. You are right. A component is...

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

I'm setting up a demo for two asterisk machines. One will be a central Asterisk server which will handle everything already in VoIP (office-like functions plus agents functionality). The second Asterisk box will be used strictly as a VoIP gateway to the first server. The gateway server will have 4 T1s connected to it and what I was thinking on doing was the following: in

I just installed a TE410P on a Debian Sarge system running kernel 2.6.11-1-686-smp. Zaptel and Asterisk seem to be working fine. However, I have a couple of problems with the TE410P and Zaptel. First, the TE410P is showing me red alarms on 2 of the 4 T1s. This board (the TE410P) was just moved from another machine running REL3 and all 4 T1s were working there. I don't know why only

I'm noticing by watching the CLI that my inbound calls coming via T1s on a TE410P are using GSM codec. Why wouldn't it use ULAW as default? How can I make it use ULAW as default? Thanks, Daniel

I just came across http://www.voip-info.org/tiki-index.php? page=Asterisk%20TDMoE and seemed very interesting. It prompted me to question whether it would be more efficient to do TDMoE or IAX2. The application is very simple. I have two asterisk boxes. One is strictly a gateway to the PSTN using multiple T1s. The second is where all my extensions and dialplan are defined (the workhorse for

Can anyone provide more information on switch or point me to where I can find more about it? The only I've been able to find on the wiki is: http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers and towards the bottom of (section Forwarding to another Asterisk): http://www.voip-info.org/wiki-Asterisk+config+extensions.conf Some of the questions I have are: 1) If I have an