similar to: Asterisk 1.6 pbx_lua not creating any contexts

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

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.

one under my default context at extention.conf. And what is [pbx_config]? Thanks Eyal -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, June 25, 2010 4:05 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Is there a default dial plan that is not in

Hello. I successfully installed lua and use lua_pbx with my dialplan, but I need to use mysql. I installed luasql.mysql and without extensions.lua it work fine (woth external scripts) I tested it with my little script: ______________________________________________________________ function mysqltest() local driver = require "luasql.mysql" local env = assert(driver.mysql()) local con

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hi I have asterisk 1.8.18 with freepbx 2.10.1.9. I get an asterisk crash occasionally with the followingerror. It always seems to happen while paging. 16 spa508g phones 1 snom pa1 paging amp Kelly == Extension Changed 1101[ext-paging] new state Idle for Notify User 101 pbx*CLI> *** glibc detected *** /usr/sbin/asterisk: malloc(): smallbin

-----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':

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

Hi, How does * read and process the extension.conf file?? The reason I ask is that I think it probably has a very large impact on how the calls are routed and processed by the system especially when it comes to least cost routing.. Let me explain...with an example.. I am using the * Devkit to get to grips with the system, so I have and X100P (Zap/1) and and S100U (Zap/2).. Below is my

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

So, incoming calls on zap work just as I expect them - an intro is played, the caller hits 1 for sale 2 for support or dials an extension. I'm using the privacy option for all extensions. When calls come in from zap, they caller is played the priv-recordintro recording, they say their name, and everything happens normally from there on out. However, when the call comes in from sip and

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]

Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? Thanks Bruce

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

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.14 and 1.6.1.5. Asterisk 1.6.0.14 and 1.6.1.5 are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 1.6.0.14 is the first full, non-security release since 1.6.0.10. The release candidate 1.6.0.11-rc1 was redone as 1.6.0.14-rc1 (which this release has been created

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.14 and 1.6.1.5. Asterisk 1.6.0.14 and 1.6.1.5 are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 1.6.0.14 is the first full, non-security release since 1.6.0.10. The release candidate 1.6.0.11-rc1 was redone as 1.6.0.14-rc1 (which this release has been created

Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING, TWILIO)). It does not work and NO error message in CLI. I have also tried http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I first emailed this group, but that does not seem to work either. Here is my log: [Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also