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WOW! Just searched the forum and found this: http://pbxinaflash.com/community/index.php?threads/sql-lite-causing-asterisk-to-crash.14134/

FYI: sqlite writing cdr is causing crashing on PAIF-green under heavy traffic Digium say res_config_sqlite is not supported (actually they say res_config_mysql is also not supported) Also turns out by default all three cdr backends are enabled: csv mysql sqlite Maybe a good idea to disable...

1) Sad to see it go. 2) Thanks for making it an easy turn up/ tear down testing environment. 3) I believe this doc with diagram will help those who want to clone it for themselves: http://docs.aws.amazon.com/AWSEC2/latest/UserGuide/creating-an-ami-ebs.html Summary: have an instance based on...

Same here: PIAF Purple (1.8) amazon AMI Fail2Ban is not installed - I checked it is not there. By the way on PIAF Green (11) amazon AMI - Fail2Ban is OK.

SOLVED. Cause: Looks like it is not implemented in AGI. Workaround: What works - is a dialplan command - for example: SayNumber(33,m) so here is a workaround: $AGI->exec("SayNumber","33|m"); or $AGI->exec("Exec","SayNumber(33|m)");

Hi, can someone help re: asterisk AGI male/female in another language? Asterisk ver 1.4.33, If I try this: [SAY NUMBER 123 ""] - it counts in female gender. If I try this: [SAY NUMBER 123 "" "m"] - I get an error: 520-Invalid command syntax. Proper usage follows:

Summary: This: Set(CHANNEL(language)=he) somehow has a strange effect: == Auto fallthrough, channel 'SIP/212.150.88.21-00000011' status is 'UNKNOWN' Details: inside extensions_custom.conf I added: [from-pstn-custom] exten => _.,1,Noop(*****from-pstn-custom*********)...

thanks WardMundy Do you know if/when this has been resolved/logged My ver: ******************************************************************* * Running Asterisk Version : Asterisk 1.4.21.2 * Asterisk Source Version : 1.4.21.2 * Zaptel Source Version : 1.4.11 * Libpri Source Version ...

Restart clears them. But shortly run up new ones again. Maybe it has to do with the way I have setup IVR? For a couple selections I have set up extentions with followme to redirect to outside numbers. "dial 1 to call 1-800-1111111; dial 2 to call 1-800-2222222..." It seems that all...

No. That is just the SIP trunk ITSP.

FYI: PBX in a Flash Forum » vBulletin Message vBulletin Message The server is too busy at the moment. Please try again later.

Anyone, please?

System status is showing: Total Active Calls 10 Internal calls 5 External calls 5 Total Active channels 20 but I dont think there are any at the moment. My ver is FreePBX 2.5.0rc3.0 Please help troubleshoot/diagnose wassup? Thanks! pbx*CLI> sip show channels Peer...

Admin and reports tabs work with admin/admin. Panel is asking for user/pass for restricted area and I could not login. What is the default password for panel?

Can end-users self-administer follow-me by phone? I know they can self-administer by web (if they have access), but can they do it with DTMFs by phone? Thanks, KW

I am doing a project with PIAF. Need to have a go-to channel for support/consulting. Remote support is fine for now. Please advise: Where I can go to get help? Cost? Thanks, KW

re: improving quality on VMware attempting to improve sound quality issues (choppy/skip/stutter/slowspeeach/fastspeech) Given: - plenty hardware - PIAF on VMware server ver 1.0.6 - vm optimized kernel - kernel /vmlinuz-2.6.18-53.1.4.el5vm ro root=/dev/VolGroup00/LogVol00 acpi=off noapic...