Hi,

I have FOP2 running on Centos 5 and it works well for 3+ years.

I'm in the process of setting up a new PBX server, which is a Centos 6.5 VM. I have installed fop2-2.28-centos-x86_64.tgz.

Upon running the server binary, I get the following error:

[vagrant@localhost fop2]$ ./fop2_server --help
Can't locate PAR.pm in @INC (@INC contains: /usr/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-multi /usr/lib64/perl5/site_perl/5.8.7/x86_64-linux-thread-multi /usr/lib64/perl5/site_perl/5.8.6/x86_64-linux-thread-multi /usr/lib64/perl5/site_perl/5.8.5/x86_64-linux-thread-multi /usr/lib/perl5/site_perl/5.8.8 /usr/lib/perl5/site_perl /usr/lib64/perl5/vendor_perl/5.8.8/x86_64-linux-thread-multi /usr/lib64/perl5/vendor_perl/5.8.7/x86_64-linux-thread-multi /usr/lib64/perl5/vendor_perl/5.8.6/x86_64-linux-thread-multi /usr/lib64/perl5/vendor_perl/5.8.5/x86_64-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.8 /usr/lib/perl5/vendor_perl /usr/lib64/perl5/5.8.8/x86_64-linux-thread-multi /usr/lib/perl5/5.8.8 .) at -e line 942.

I have version 5.10.1 of perl:

[vagrant@localhost fop2]$ perl --version
This is perl, v5.10.1 (*) built for x86_64-linux-thread-multi

I have found various posts in the forum referring to http://www.fop2.com/documentation-faq.php , but this doesn't seem to offer any details regarding Centos 6.

Any help would be much appreciated,

Thanks.

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.

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.

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

Sure..

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

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

Yes, the originate command is sent when the Dial button is clicked, but no originate command is sent by fop2_server to asterisk when the dialbox is used.

The originate command I first posted is a result of clicking on the Dial button, when the dialbox is used, nothing is sent to the asterisk server.

Now I originate a call by clicking on the dial box. This works - the call is originated successfully. Below you can see a portion of the full log

[Apr 8 12:22:09] DEBUG[29455] manager.c: Manager received command 'Originate'
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Asked to create a SIP channel with formats: 0x40 (slin)
[Apr 8 12:22:09] VERBOSE[11496] netsock.c: == Using SIP RTP TOS bits 184
[Apr 8 12:22:09] VERBOSE[11496] netsock.c: == Using SIP RTP CoS mark 5
[Apr 8 12:22:09] VERBOSE[11496] netsock.c: == Using UDPTL TOS bits 184
[Apr 8 12:22:09] VERBOSE[11496] netsock.c: == Using UDPTL CoS mark 5
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Allocating new SIP dialog for <!-- e --><a href="mailto:[email protected]">[email protected]</a><!-- e --> - INVITE (With RTP)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Setting NAT on RTP to On
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Setting NAT on UDPTL to On
[Apr 8 12:22:09] DEBUG[11496] acl.c: Found IP address for this socket
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.30.10:5060
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** Our native formats are 0x4 (ulaw)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** Joint capabilities are 0x0 (nothing)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** Our capabilities are 0x4 (ulaw)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: *** Our preferred formats from the incoming channel are 0x40 (slin)
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: This channel will not be able to handle video.
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Outgoing Call for 14502
[Apr 8 12:22:09] DEBUG[11496] chan_sip.c: Updating call counter for outgoing call
[Apr 8 12:22:09] DEBUG[9984] devicestate.c: No provider found, checking channel drivers for SIP - 14502
[Apr 8 12:22:09] DEBUG[9984] chan_sip.c: Checking device state for peer 14502
[Apr 8 12:22:09] DEBUG[9984] devicestate.c: Changing state for SIP/14502 - state 6 (Ringing)

Now, if I type the extension into the dialbox and hit ENTER, I cannot find anything in the full log relating to this request - no mention of and 'Originate' command or anything like that. The only output I see in the logs available to me are the dial msg tag being sent by the web client (seen in firebug), and the msg being received by fop2_server when running it at debuglevel 15.

So FOP2 succeeds in originating the call when the Dial button is clicked, but is unable to originate a call when the dialbox is used.

I am running FOP 2.2 against asterisk 1.6.2.3 managed by FreePBX 2.8.0.2.

After running for a while, with not much usage as we are only testing at present, the dialbox has stopped working completely. I can successfully originate a call by selecting an extension, then clicking on the Dial button. If I enter the same extension into the dialbox and hit ENTER, nothing happens.

Below is an excerpt of the output from the fop2_server at debuglevel 15 when using the Dial button:

10.0.28.248 <= <msg data="6|originate|8|3779e1ed97121b000bd42c3d75ced8b4" />

127.0.0.1 -> Action: Originate
127.0.0.1 -> Channel: SIP/14502
127.0.0.1 -> Exten: 4008
127.0.0.1 -> Context: from-internal
127.0.0.1 -> Priority: 1
127.0.0.1 -> CallerID: Reception 1 <4006>
127.0.0.1 -> Async: True

127.0.0.1 <- Response: Success
127.0.0.1 <- Message: Originate successfully queued
127.0.0.1 <- Server: 0

.....

10.0.28.248 => { 'btn': '6@GENERAL', 'cmd': 'settimer', 'data': '0@UP', 'slot': '1' }

10.0.28.248 => { 'btn': '6@GENERAL', 'cmd': 'state', 'data': 'RINGING', 'slot': '1' }

10.0.28.248 => { 'btn': '6@GENERAL', 'cmd': 'settext', 'data': '4006 Reception 1', 'slot': '1' }

And when using the dialbox:

10.0.28.248 <= <msg data="6|dial|4008|3779e1ed97121b000bd42c3d75ced8b4" />

The above line is the only output when using the dialbox, so it seems that the dial message is being silently dropped. I have checked the various asterisk settings - callevents=yes, read/write=all, event_mask commented out.

Thanks.

Sorry, I confused things as I'm experiencing a second issue which means that I can't test the original problem.

I'll post a new topic, and come back to this one when I can reproduce it.