It looks like that page was taken down but I fid find it saved in the Internet Archive. The link is at:
Might be out of date, but at least something to get started with!
It looks like that page was taken down but I fid find it saved in the Internet Archive. The link is at:
Might be out of date, but at least something to get started with!
It's not the dahdi_dummy, that is really not needed anymore if you use app_confbridge and not app_meetme for conferences.
It is almost certainly an admixture of your hardware and your choice of virtualization technology, as the cpu/network passed through to the VM and the transparency of that VM to the real world is your problem.
I would try using KVM (qemu) on a linux machine with VT-X/AMD-V enabled CPU and decent hardware, ensure the virtualization technology is enabled in the bios.
You could quickly test the whole shebang with:-
https://www.proxmox.com/en/downloads
OS/GUI/KVM all setup for you (4 FreePBI and a Windoze of choice easily on a NUC i5)
On my phones in the auto provision field I put: tftp://192.168.0.25
I don't know if maybe this is mandatory in the older 3 series phones
So, I guess I need to wait for PIAF to make a update available?
Thanks a lot. this saved me a world of problems.
Hi!
Try updating and if it doesn't work open a ticket at http://issues.freepbx.org.
Good luck and have a nice day!
Nick
Does anyone have any suggestions for the socket error on where I should look to resolve this problem as it is still persistent?
This behavior explains why there is no .tiff file which is mentioned in my log files.
The far side receiving fax directly to an analoge fax.
The highest value under fax configuration is 14400, I also tried 9600 for both but nothing changed.
This sounds like a communication issue which can be caused by dahdi drivers, network configuration, fax/ATA configuration, fax machine configuration, etc. Without more specifics about call paths and settings on every device being used along the way, it is almost impossible to troubleshoot this way. If the ATA and fax machine are on the same LAN, I would look at the configuration of the ATA and fax machine first. Ensure the fax machine is configured to negotiate at 14400 or lower also.
This seems to be a webkit bug. If you google the issue "safari adds .html" you will see complaints going back 7 years. This does not happen in non webkit browsers. Where most browsers have some sanity checking to figure out what the file is webkit relies on mime types. I have filed a bug for this and pushed a fix/branch against it.
Yeah I am thinking that too. I have a case started with Digium. I want to tread lightly here since this is a large Dr office.
If I understood you correctly, you want to execute the script when a call arrives at the IVR, right?
The way I would do it, it's send the "Inbound Call" to a custom extension where it would execute the script and then forward the call to the IVR.
extension_custom.conf
exten => 1000,1,Answer()
same => n,GotoIf($["${CDR(disposition)}" = "ANSWERED"]?begin)
same => n,AGI(DIPControlLlamadas2.agi)
same => n,Goto(from-did-direct-ivr,s,1)
Something like that.
Inbound -> Custom Extension -> IVR
Don't forget to register your extension in the "Admin" > "Custom Extensions".
Sorry but now I am confused.
I like to get an incoming fax as an E-Mail so I have the following sequence of events:
any sending fax -- via ISDN ---> my Beronet Card ---> FreePBX Inbount Route for this number ---> Fax Reciptient -- create PDF ---> E-Mail to defined address of the User
In this scenario I cannot take influence about the ATA or the fax machine, can I?
The sending of faxes is irrelevant at the moment for me.
My configurations are linked in my first post. Nothing more is configured by me, cause in my mind there is nothing more to configure.
I am no guru in this section of technology, so please explain if I am completely wrong.
Network configuration:
_________###############
_________#_____________#
_________#____FreePBX__#
___ISDN__#_____________#
(ouside)_#_______________#__LAN
----->____#____Beronet____#--------------> PC
________#_______________#
________#_______________#
________#################
FreePBX:
IP: 192.168.99.201
Netmask: 255.255.0.0
Gateway: 192.168.0.1
PC:
IP: 192.168.10.10
Netmask: 255.255.0.0
Gateway: 192.168.0.1
Beronet:
IP: 192.168.99.202
Netmask: 255.255.0.0
Gateway: 192.168.0.1
Not sure if works for your specific model, but you may need to disable the SIP ALG.
Rodrigo's code will get you half way there, but it won't work as written. You don't want to use a Custom Extension, you need to use a Custom Destination.
Add a block of code to extensions_custom.conf that looks similar to this:
[agi-trigger]
exten => s,1,noop(entering context [agi-trigger] in extensions_custom.conf)
exten => s,n,AGI(DIPControlLlamadas2.agi)
exten => s,n,Return
In FreePBX, Admin, Custom Destinations, create a new Custom Destination that references the code block above like:
agi-trigger,s,1
Select the return option, and choose the desired destination (IVR or whatever)
Now instead of sending callers to the IVR, send them to the newly created custom destination. This will execute your AGI then send them to the IVR.
You have enabled the UCP Node server. But either you have the ports blocked for it or it's not running.
Thanks guys! I ended up making a custom extension with
local/2XX@from-internal
and it works well and easily for adding to ring groups etc.
I guess there is no best practice or would it be the hashtag #.?
John
You have invalid data in your /etc/asterisk/voicemail.conf file
if you look into your asterisk logs, do you see any messages related to a communications timeout? If this is not the problem, it is likely a fax negotiation issue when the fax tones start.
Thank you Andrew. I found the invalid data and the module is now loaded.
However, secondary problem now. I get a fast busy whenever trying to access the directory, whether from IVR or an inbound route. Attached is test call from DID 6042148511: