similar to: Opening for Sr. Ruby On Rails Developers

I am using rails 3. I have problem in out put so please tell me my mistake routes ------- entermatch ''/view_article/:user_id/:article_id'' => ''articles#view_article'', :as => :view_article resources :users do resources :articles do resources :comments end end article controller ------------------ def view_article @user =

Hello, I am Ainish Dave from Ahmedabad, India. I am currently pursuing my masters level degree with background in computer science. I browsed through the list of organizations on the GSoC program page and Xapian was the one which closely matched with my interest and background. I had some queries: 1) I am interested in the project 'Clustering of Search Results'. I have followed the

Seems like this is the best idea (challenge-response), a callback. No matter the callerid, you don't know where the caller is. But if you place a call BACK to the callerid, it's going to go to the destination. Then you either need the phone to be answered, or the phone to be answered and and the challenge entered. Adam Goldberg AGP, LLC +1-202-507-9900 -----Original Message-----

Hello, I'm trying to get Super MNC running under Wine with Steam, I can start the game up perfectly fine and join training matches with other players, but I can't join a non-training match, when I try to it puts me in a queue which I never leave. Here's the ouput which includes Steam's output also: http://pastebin.com/LDTEgbZz Thanks.

Hi, I have a doubt regarding MDCT window size for OggVorbis decoder. Can the MDCT window size vary from 64 samples to 8192 samples? Specification says that "legal frame sizes are powers of two from 64 to 8192 samples" . Is this same as MDCT window size? Please help me out.. Regards, shikha <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

On Wed, 10 May 2017, J Montoya or A J Stiles wrote: > Presumably your staff carry mobile phones. What about an app that gets > the ID of the cell tower to which it is connected, and passes it and the > SIM number in a HTTP request to a server you control? The problem is that they are supposed to use the 'site landline' to confirm presence -- not their cell phone with the

Hi everybody i m ashish here now i m not using this mail more so pl.mail me on abbarot at gmail.com Thank you Barot Ashish Jr.System Administrator, Lubi Electronics, Ahmedabad. __________________________________ Yahoo! Messenger Show us what our next emoticon should look like. Join the fun. http://www.advision.webevents.yahoo.com/emoticontest

Hi. I am trying to get Samba 3.0.4, from tarball, working with Fedora Core 1. The problem is that I can't get LDAP authentication with either ssl or start_tls support. I tried compiling ldap on a plain install of fedora. I also tried compiling it after installing openssl-0.9.7d and openldap 2.2.11 clients and libraries from tarball. My ldap configuration of samba is as follows:

hello , all i want free g729 codec for asterisk i tried so many moduless on asterisk.hosting.lv but cant find any related codec to my machine i cant understand where to start my asterisk version is 1.6.0.5 and following are output of cpu cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 23 model name : Intel(R) Core(TM)2 Duo CPU

Dear Members, I have been searching for a package in R which can handle multiple seasonality suggested by taylor(2003). It will be great help if anybody has used this on R before (i.e. which package). Thanks in Advance. Best Regards Atul Malik [[alternative HTML version deleted]]

Hi, If I take up the task of 'clustering the search results' and I am opting for the new method to cluster the results, then what will be the approximate amount of data that will be clustered? So appropriate clustering mechanism can be thought of according to that. Thanks & regards, Ainish On Sat, Mar 19, 2016 at 5:37 PM, James Aylett <james-xapian at tartarus.org> wrote:

I have this code in a dial plan: exten => _417XX,n,GotoIf($["${CALLERID(num)}" > "SIP/41799"]?notfromlocal) exten => _417XX,n,GotoIf($["${CALLERID(num)}" < "SIP/41700"]?notfromlocal) The value of "${CALLERID(num)}" appears to be "SIP/41712-00000181" -- Executing [41720 at from-internal:5]

Hey guys, I want to send an email via ROR and I have put this code at the very end in the config/environment.rb file- ActionMailer::Base.delivery_method = :smtp ActionMailer::Base.perform_deliveries = true ActionMailer::Base.logger = nil ActionMailer::Base.raise_delivery_errors = true ActionMailer::Base.smtp_settings = { :address => "[my smtp server]", :port => 25, :domain

I am stumped In features.conf,I programmed this [applicationmap] Answer0 => 0,self/both,Macro,nway_start But do I pass an argument or parameter to my macro? I tried Answer0 => 0,self/both,Macro,nway_start^0 Answer0 => 0,self/both,Macro,nway_start,0 but the usuar variable ${ARG1} is empty in my dialplan. The issue is that my macro needs to know what key was pressed. -------------- next

Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

Trying to figure out the best way to pull an active call out of a queue by unique id and put it on hold. I don't want to put it on hold on the agent's phone but I want it to be pulled away from the agent's phone and into Asterisk limbo somewhere. Shortly after I want to pull the same call out of limbo and redirect it back to either the same agent or another. I was thinking about call

Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At first, DIALSTATUS doesn't return any status. How to now if a call in queue has been answered or if caller just hangup? Second, how to deal with timeout, I have strange behaviors. If I put timeout=60 in queue.conf and I call the queue passing also 60 as timeout value,

hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same