similar to: cannot run agi scripts

On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'

Topic 6: RE:unable to execute command hi there i tried to execute the command as suggest like exten => 1987,1,Playback(posix-restarting) exten => 1987,2,wait(1) exten => 1987,3,System(/usr/bin/python /home/docas/Desktop/mess1.py) exten=> 1987,4,Hangup it still doesn't work,now it does it give unable to execute command but it doesn't reach the system command it just

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get

Hi Shakeel, I had the same problem building app_swift (1.6..) myself and searched the web far-and-wide for a solution. I eventually contacted Darren Sessions -- who was maintaining that plug-in -- about a month ago. He was involved in another project and said he might be able get to it after a few weeks. But, since then, his website http://www.darrensessions.com/ has gone out of comission. I

Hi everyone I am having problems with my call queue We currently run a customer care call center which has attendants login during the daytime. Customers who call the 'customer care line (a specific number) always get routed to the cutomer care queue (called 124). After hours, staffs of the Network operating center provide customer care services for customers who call in after the last

Hi, I'm getting the following at asterisk startup. OCBC was working with 1.4 no problem, but now under 1.6 (I've tried 1.6.1, 1.6.2-beta3/beta4) I can't seem to get rid of this .... anyone? WARNING[32664]: loader.c:385 load_dynamic_module: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache

I've configured asterisk 1.4 to use imap storage for voice-mail and while I'm happy with it generally speaking it really seem to hammer the IMAP server. It appear, from the IMAP server logs that it's polling the imap server every *second* for mailbox updates for the users' voice-mail folders. Is it really necessary to do this once a second? Is this tunable anywhere? Thanx, b.

Hi all, in witch way can I put in a log file the asterisk console? I have tried with some settings in file logger.conf but the log not contain the same debug that I can see with "asterisk -rvvv". I need it in debugging purpose for tracking some bug. Thanks Enrico. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:

after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for

1.6.2: -- Executing [s at incoming-pstn-line:4] VoiceMail("DAHDI/4-1", "100 at default,u") in new stack -- <DAHDI/4-1> Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]:

Hi, I'm trying to understand an issue I'm seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then

Hi, I have just installed the fresh svn version of asterisk and when I run it I get the following errors: [Mar 4 14:19:27] WARNING[24527]: loader.c:728 load_modules: No 'modules.conf' found, no modules will be loaded. [Mar 4 14:19:27] NOTICE[24527]: manager.c:2681 init_manager: Unable to open management configuration manager.conf. Call management disabled. [Mar 4 14:19:27] NOTICE[24527]:

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

It should be as simple as editing voicemail.conf : ; Voicemail Configuration ; [general] ; Formats for writing Voicemail. Note that when using IMAP storage for ; voicemail, only the first format specified will be used. format=wav49|wav|gsm ; Who the e-mail notification should appear to come from serveremail=asterisk-voicemail ; Should the email contain the voicemail as an attachment attach=yes ;

I am trying to launch Asterisk on a different directory with the parameter 'C asterisk -vvvvvvvvvvvvvvvvvvgc -C /etc/asterisk1/asterisk.conf Parsing '/etc/asterisk1/extconfig.conf': Found Resetting translation matrix UUID system initiated Parsing /etc/asterisk1/asterisk.conf == Parsing '/etc/asterisk1/asterisk.conf': Found Not changing threadpool size since new size 0 is

Hi, I am wondering, if there is any tool available, which performs a check for suspicious entries in the dialplan. For example a non existing AGI-Script or a double assigned extension ike that: [context] exten => *100*,1,AGI(test_app.pl) ... exten => 190,1,Answer() ... exten => *100*,1,AGI(never_reached.pl) ... A "normal dialplan reload command" would echo no warning or

We want to use Nagios to monitor out puppet server so we can be notified if it goes down. We are using Fusion Passenger and Apache on Red Hat. Any suggestion for what and how to monitor? -- Thanks, Allan Marcus 505-667-5666 Allan@lanl.gov -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to

Hello, I am trying to get a repeatable build setup for asterisk. Part of doing so involves using the -C option to specify the master config file. The problem is that asterisk reads the config file location that i specify, however it still tries to read two other config files, namely: * /etc/asterisk/extconfig.conf, and * /etc/asterisk/logger.conf I have specified in my config file that the

Hi, I am trying to set up a dial plan und I have a few problems to realise some functions. The dial plan should look like this: 123,1,Answer() 123,2,Queue(1stlevel,t) 123,3,Queue(2ndlevel,t) 123,4,Queue(3rdlevel,t) 123,5,Hangup() If a member of the 1stlevel-Queue can answer the call it should be hanged up after finishing. If not, the current member answering the call should be able to transfer