Comments on: Getting started with FreePBX – Part 6 Cheap phone calls using DISA and Callback

By: matt

matt — Fri, 06 Aug 2010 14:12:47 +0000

Nice one Lee.

Thanks for the update. Matt.

By: Lee

Lee — Fri, 06 Aug 2010 12:19:39 +0000

Callback now working,

I found the answer here:

http://pbxinaflash.com/forum/showthread.php?t=7238&highlight=callback

In short the callback timeouts are incorrect on the older versions of Freepbx which means the system will hangup after 15 seconds and terminate the callback. You need to edit this file:

/var/lib/asterisk/bin/callback

And change from:

$timeout = “15000″;

$timeout = “30000″;

And it will work.

Cheers
Lee

By: Lee

Lee — Wed, 14 Jul 2010 06:18:55 +0000

Hi Matt,

At the moment I just want it to got to DISA as my client has there own server but eventually I want to set it up for all my clients through A2billing.

My A2billing version is 1.3.0

Lee

By: matt

matt — Tue, 13 Jul 2010 11:59:49 +0000

Hi Lee,

Callback has always been a pain to get working. What version of a2billing are you using?

As the call is being established it sounds like it’s not getting passed to DISA correctly (is that where the call should be going or straight through to a2billing?)

You’d probably need to turn up the logging level to see what’s going on.

Matt

By: Lee

Lee — Tue, 13 Jul 2010 11:47:04 +0000

Hey Matt,

Great tutorial as ever.

Have been trying to set this up for a client and all steps work up until the final part.

When I receive the call back the system terminates the call when I answer.

Any ideas?

Cheers
Lee

By: matt

matt — Tue, 06 Jul 2010 23:35:02 +0000

Sure you can do that. You just need to create inbound routes with the caller id set and then forward those calls to your DISA.

By: John

John — Tue, 06 Jul 2010 15:24:29 +0000

I just switched to freepbx from text asterisk, very nice guide. Before my change I had setup disa with caller-id authentication, and this was only for family members and consisted of three numbers. Is there a way to duplicate this in freepx?

By: zenny

zenny — Mon, 15 Feb 2010 09:21:17 +0000

To my post above, let me copy part of the output of the ‘asteriks -rvvv’ command on console.

– Executing DISA(“SIP/66.54.140.46-09c9a028″, “/etc/asterisk/disa-2.conf”) in new stack
— Executing Answer(“SIP/66.54.140.46-09c9a028″, “”) in new stack
— Executing Wait(“SIP/66.54.140.46-09c9a028″, “1″) in new stack
— Executing AGI(“SIP/66.54.140.46-09c9a028″, “directory||from-did-direct|l”) in new stack
— Launched AGI Script /var/lib/asterisk/agi-bin/directory
directory||from-did-direct|l: Notice: vm-context not specified. Using ‘default’
— Playing ‘dir-intro’ (language ‘en’)
— Playing ‘dir-intro’ (language ‘en’)
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on ‘SIP/204-09bf3a30′ in macro ‘dialout-trunk’
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on ‘SIP/204-09bf3a30′
— Executing Macro(“SIP/204-09bf3a30″, “hangupcall|”) in new stack

It shows that instead of pointing me to the new dialtone to dial outside, it seems launching ‘Launched AGI Script /var/lib/asterisk/agi-bin/directory’. Help appreciated!!!

By: zenny

zenny — Mon, 15 Feb 2010 09:01:00 +0000

Since one of my ipkall number expired, I requested for a new one and I configured exactly as stated here, but when I call the DID number, I get the response to enter my PIN and #, and immediately after I get an announcement to asterisk user directory instead of the dial tone to call outside. Where did I go wrong? Any pointer? (it was working well earlier with the similar setup). thanks,

By: Av

Av — Fri, 05 Feb 2010 21:45:38 +0000

Thanks alot, it help me. you ar ethe best