General Help

A community support forum for FreePBX users.


sandovalg's picture

So I am receiving the above error with a single license FFA on FreePBX 2.8.1 and Asterisk Here is the full error. Please forgive any information that I may be leaving out and let me know if there is a better way to track down this problem. Attached is the error log. Not sure what else to include, after scouring the forum's I still have no idea concerning my next step.

[2011-08-02 15:53:34] VERBOSE[17280] app_dial.c: -- SIP/4001-0000000f is ringing


Paid Support

AshCan's picture

It looks like the paid support portal is been upgraded.

I'm trying to buy a support credit and it is giving a credit card error. (Credit card is not valid)

I have tried 2 different cards number and details. (AMEX $ VISA)

Please suggest how to proceed ASAP.


How can I limit the amount of time a person can leave a voicemail message for?

VoIPTek's picture

Hello All,

I see a voicemail hit 750M on one of the servers, so I checked through the different settings via the web gui but found nothing allowing me to control how many minutes a person can leave a voicemail for before disconnecting the call.

Any suggestions?



Sometimes a phone i a queue does not ring

jmaurib's picture


I've been experiencing this problem for a while and even though . I have a FreePBX (no v 2.9) running in a Asterisk (now 1.4.42). From now an then one of the phones in a queue does not ring.
I've been able to see that when this happens I get a manager unable to connect message, something like that:

VERBOSE[15276] logger.c: == Connect attempt from '' unable to authenticate

and of course the macro-dial returns:


restore from backup

darrainw's picture

I am using 2.2.12 ce, and am able to restore to a new server with the same version. However, the telephones are not able to call each other. We are kinda stuck with this version for now as we do not want to re create every extension in a newer version. Thanks for the help.


Outgoing DAHDI calls not showing CPN

awdavis214's picture

I have a Digium TE420 with 3 ports configured as PRIs and 1 as E&M wink. Our analog lines use the E&M trunks to make outgoing calls. The calls are being processed fine but no CPN is being displayed. Does anyone have any ideas?



#include /etc/asterisk/dahdi-channels.conf


; Autogenerated by /usr/sbin/dahdi_genconf on Wed Nov 3 00:02:56 2010
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)


voicemail calls not hanging up

decaturjohn's picture

I've been dealing with this problem for a while now and hoping I can find a fix for it. I have a problem with voicemail calls hanging and asterisk not actually delivering the vm to the end user until I reboot the phone server. In some cases it will hang for a month and I wouldnt know about it until the end user comes to me about a voicemail they just received from a month ago, despite checking that vm box everyday. So it appears that the actual recording of the voicemail works, but asterisk cant deliver the vm until the call is terminated at the phone server.


What O/S and Asterisk version is the FreePBX Distro + Remote Access, & vs AsteriskNOW?'s picture


Three questions in one to satiate my curiosity. If you can help much appreciated!

Just wondering what the Linux O/S is that's bundled with the FreePBX distro?
I can't find where it says.

Is the Asterisk version 1.8.2?
Again I can't see where it says but was taking a clue from the ISO file name.

Does anyone know if TeamViewer can be installed on this or some other application that readily bypasses firewalls without port forwarding for easy continuous remote support?