Dial box not working

[nicolas]

Ok, so I misunderstood your 1st post. You will have to increase the debug level to 511 when originating from the dialbox, you might also want to be sure you are using fop 2.20 final and not the beta. Are you on user & device mode in freepbx ?

[kevinmusker]

Yes, we are running in device/user mode and a registered copy of FOP2 version 2.20 final.

I think we are getting somewhere now. This is the output from debuglevel 511:

** MAIN AMI event received...
** MAIN Processing command received from flash clients...

192.168.180.106 <= <msg data="6|dial|4007|c4dbe7308d413f2c15365e7470249bde" />

-- PROCESS_FLASH_COMMAND origen 6 accion dial destino 4007

-- PROCESS_FLASH_COMMAND password c4dbe7308d413f2c15365e7470249bde

VALIDAR USUARIO 4006

VALIDAR USUARIO 4006 OK con clave regular (192.168.180.106)

Validation ok, have dial permissions

Not a reference at all

DIAL failed because origin channel SIP/14502 is not blessed

DIAL

The above is when signing into FOP2 as extension 4006, which is fixed to device id 14502.

Interestingly, if I try the same thing when logged into FOP as an extension that is not fixed to a device, but logged in to an adhoc device, it works:

** MAIN AMI event received...
** MAIN Processing command received from flash clients...

192.168.180.106 <= <msg data="8|dial|4007|9bdefc597220a3c8710d6be15ac03809" />

-- PROCESS_FLASH_COMMAND origen 8 accion dial destino 4007

-- PROCESS_FLASH_COMMAND password 9bdefc597220a3c8710d6be15ac03809

VALIDAR USUARIO 4008

VALIDAR USUARIO 4008 OK con clave regular (192.168.180.106)

Validation ok, have dial permissions

It's blessed into class Extension

DIAL
Action: Originate
Channel: SIP/14501
Exten: 4007
Context: from-internal
Priority: 1

[nicolas]

Just to be sure, can you tell me the md5sum for /usr/local/fop2/fop2_server ?

md5sum /usr/local/fop2/fop2_server

[kevinmusker]

Sure..

[root@pbx1 fop2]# md5sum fop2_server
b20267e789e235d02267b33c3e43f5fe fop2_server
[root@pbx1 fop2]# ./fop2_server -v
fop2_server version 2.20

[kevinmusker]

Hmm, the fixed/adhoc device correlation may have just been coincidental - the dialbox is not working for all extensions now - fixed or adhoc.

Same error as before:

DIAL failed because origin channel SIP/14501 is not blessed

[nicolas]

Do you have "all" permissions both in read/write in /etc/asterisk/manager.conf ?

[kevinmusker]

We have a dedicated user defined with read=all, write=all.

I also checked fop2.cfg for an event_mask line - there is one, but it is commented out.

I tried restarting everything (fop2, asterisk, ctrl-F5 in browser) to make sure that nothing had got in a weird state, but it didn't help.

[nicolas]

I will have to set a system in user & device mode for testing this and see if I can reproduce the issue, or you can try to catch me on the live help and provide remote access to your server so we can work on there directly.

In the meantime, there is probably a way to fix this problem using the originatechannel configuration parameter. If you have the fop2admin module installed, go to Tools - FOP2 Buttons, and set the originatechannel of one of your users to Local/XXXX@from-internal where XXXX is the user/extension number. Then try again.

Best regards,

[kevinmusker]

Thanks Nicolás.

I tried setting the originate line, but it did not work. The error message is essentially the same:

DIAL failed because origin channel LOCAL/4006@FROM-INTERNAL is not blessed

Unfortunately I cannot provide you with remote access. As a developer, I understand how difficult it can be to diagnose a problem without access to the system with said problem. The system I am working on will become our live server soon, so has a lot of user information.

If there is anything I can send you directly - database dumps, config files, etc, I'd be happy to provide them. I can also try anything else you might suggest.

Thanks for your continued help.

[macdsaint]

Hi,

Im also experiencing this with the latest version. Im using asterisk 1.4.22.
I just downloaded and purchased the basic version.

When i hit enter on the Dial textbox, it does not react nor any debug messages to the asterisk cli is popping out

Appreciate any assistance on this matter

Thanks

[nicolas]

Are you using device & user mode in freepbx? Or custom button configuration? If the later, please post a snippet of your button config. You can start fop2_server in debug mode and look at the output looking for clues.