similar to: PBX Lua with Asterisk ODBC

Hi, After compiling a installing asterisk 1.8.2.3 I wanted to play with lua but I noticed that extensions created in extensions.lua was not being registered with asterisk. uga1*CLI> dialplan show [ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ] 's' => 1. NoOp() [app_queue] [ Context

Please could someone advise good manual for using lua for asterisk dialplan. There is not much docu about it. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110218/42642787/attachment.htm>

Hello I'm taking a look at how to write scripts to be called from the dialplan, and saw pbx_lua mentioned. I'd like to know more about this feature, such as what the difference is with just calling the Lua interpreter through AGI (same difference as between php-cgi and mod_php?), whether it's production-ready, etc. Thank you for any help.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Hi I have searched a bit for information regarding the status on the dialplan in lua (pbx_lua.so). I know that 'hint' won't work and has to be put in the regular extensions.conf/ael. Are there any other considerations? Features that are not implemented, or known stability issues? Thanks, Tommy Botten Jensen -----BEGIN PGP

Hi to all! I'm trying to create a context for integration with extensions.lua and libsql.mysql, but I'm not getting to run. When I reload the module pbx_lua.so the following error appears: [Feb 24 16:59:29] ERROR[30749]: pbx_lua.c:1249 exec: Error executing lua extension: error loading module 'luasql.mysql' from file '/usr/lib/lua/5.1/luasql/mysql.so':

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Hi, I have defined a dialplan in lua and now would like to use "dial" with the macro M to implement some logic, when the callee-channel gets created. Working old style would be (extensions.conf) [default] exten => _X,1,dial(SIP/1,,M(mymacro^parameter)) [macro-mymacro] exten => s,1,verbose(${ARG1}) How to implement the same functionality using pbx_lua? Details: Asterisk 11.7

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 All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

Hello everyone. I need a quick help on how to capture who answered the call with agi. Here is an example: -- Zap/32-1 is ringing -- Zap/33-1 is ringing -- Zap/34-1 is ringing -- Zap/35-1 is ringing -- SIP/operator1-e77f answered Zap/23-1 So how can I capture this value and write it to mysql? I already have this:

I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100726/c1463f31/attachment.htm

Howdy, all. I'm trying to use pbx_lua as included in Asterisk 1.6 -- but while it correctly reports an error on startup (but not reload!) if extensions.lua does not exist, it doesn't appear to actually create any contexts. I'm testing in a very minimal configuration with autoload turned off; "module show" shows only chan_sip, pbx_lua, and app_dial. "dialplan show"

How does one go about accessing gosub arguments from Asterisk in extensions.lua? For example, I have the following in extensions.conf: exten => 1000,1,Wait(1) exten => 1000,n,Gosub(functions,mytest,1("123")) exten => 1000,n,Hangup And then the following in extensions.lua: extensions = { functions = { ["mytest"] = function()

Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal

Stop advertising. Le 26 sept. 2010 09:46, "Faisal Hanif" <faisal at vopium.com> a ?crit : Hi Abdul-Basit, If you need only different intervals of billing you can easily do it using any AGI as we are doing it in Perl AGIs using post call billing. But if you need realtime billing then the most stable and flexible option is to use FastAGI+ AMI. I have tested it in JAVA and it

Hello, we are using hardware array and its vendor recommends the following setting in /etc/system: set sd:sd_max_throttle = <value> or set ssd:ssd_max_throttle = <value> Is it possible to monitor *somehow* whether the variable becomes sort of bottleneck ? Or how its value influences io traffic ? Regards przemol

Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. Thanks

Hi All, Does anyone know what the current status is of the G729 codec on Solaris? According to the following link: http://www.asteriskvoipnews.com/asterisk_releases/_digium_g729_codec_now_available_forsolarissparc.html there is a version available for SPARC processor's. However, I have just had a quick look around Digium's FTP server and cannot seem to find these codecs (supported or

Hi Everyone Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell Canada. User claims that call hangup without any interferance to the phone set. Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? I checked the *"asteriskcdrb"* table and it's pretty much useless in this case as it only logs the duration and