similar to: Migration in Multiple Database

hello to all, i am getting weird syntax error while running the rake db:migrate syntax error, unexpected tSYMBEG, expecting kDO or ''{'' or ''('' Apparently i am creating an engine in my main application using ''ruby script/generate plugin'' command in rails 2.3.0 with engine having its own separate database define in database.yml of main

Guys, I''ve run across an interesting scenario with my migrations, and I hope you might share some insight. Let''s say I have 3 migrations and two Models (Person and Car): 001_initial_schema.rb 002_data_load.rb 003_add_car_association.rb The first creates the initial schema...the second loads default data using model objects of Person to do so, and the third creates a new

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there ..... ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Suppose I have fun.h as: static void fun() { int a =10; } Now I have two files foo.c and goo.c as foo.c : #include "fun.h" void foo() { fun(); } goo.c: #include "fun.h" void goo() { fun(); } I get .bc files for foo.c and foo.bc through clang. Now I run llvm-ld with -disable-opt for foo.bc and goo.bc. In the resulting .bc files, one of the

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

Hi Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]:

Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. *AMI login:- * *login.php* <?php $socket = fsockopen("127.0.0.1","5038",

hi list, please help me how to know how many calls are on hold..... -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110517/09cbc325/attachment.htm>

Hi Team, I have installed asterisk-12 Beta but when I try to asterisk start then get below issue. *[root at cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory [root at cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#* -- Thanks and regards Virendra Bhati +91-9718500594

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

Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then

Hi List, When we make calls into asterisk with the help of our mobile, landline number, Cisco 79XX series then we didn't able to here any IVR which is playing into asterisk server. But when we dial from SIP softphone then all is working fine and we are able to here the IVR sound files. What is the problem in this case please help me.. -- ----- Thanks and regards Virendra Bhati

Hi List, I want to set my caller ID and name with asterisk. So that when I make outgoing calls then destination end will see my name with number. from asterisk end I set both the things into dialplan. --------------- -------------- exten => _X.,n,Set(CALLERID(num)=9172341457) exten => _X.,n,Set(CALLERID(name)="Virendra Bhati") But when call reach to destination number then only

Hi List, I am trying to get DTMF into conference room. for conference I am using Konference module. Konference don't have an option of DTMF gets. Is there any way by which I can get DTMF within conference room? ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

How to make outgoing calls from DID and what is theway to get incoming calls from DID. -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110526/3f19091d/attachment.htm>

Hi List, I have stupid question but I want to know it. Why we use the PRI insted of BRI ? Just for the sake of number of lines or any thing else ? And why SIP is used for making calls rather then IAX? Even we know IAX takes 1 channel for making calls? ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Reader -------------- next part -------------- An HTML attachment was

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf* [incoming] exten => 6000,1,Answer exten =>

On Mon, Jun 8, 2009 at 3:25 AM, <Vasudev.Negi at microchip.com> wrote: > > I get .bc files for foo.c and foo.bc through clang. Now I run llvm-ld > with -disable-opt for foo.bc and goo.bc. In the resulting .bc files, one > of the two functions fun, is renamed to fun<number>. There are two > llvm.dbg.subprogram descriptors created for the two fun functions. Both > have