FOP2 Community Forum

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:183e2ef558e7ac96476632d574f6e545@10.0.30.10">183e2ef558e7ac96476632d574f6e545@10.0.30.10</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.

Hi,

We have a dedicated AMI user set up with 'all' for read and write.

Running fop2_server with debuglevel 15 gives the following output when using the dial box:

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

When using the dial button, I get this

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' }

It looks like fop2_server may be dropping the dial request without contacting asterisk at all. Its also the same if I try to dial a non-extension number, such as an external line.

Hi Nicolás,

I checked, and we have callevents=yes specified in sip_general_custom.conf. I also checked the rest of the asterisk config files to make sure that we weren't setting it to 'no' somewhere else.

Interestingly, the dial box has stopped working completely now. When I enter an extension into the dial box and hit Enter, I can see in firebug that the flash command is being sent, but nothing happens after that - no call is originated. When running fop2_server with devuglevel 2, I don't see any output for the dialbox commands any more.

Thanks for your help.

We are experiencing an issue with the dialbox on FOP2. When not in a call, the dialbox works as expected and originates a call. When in an active call, using the dialbox fails to initiate a transfer. Clicking the transfer button works as expected.

Using the javascript console, I see the command being sent via the flash object:

flash send <msg data="6|dial|4008|4e3ff625440add245d32b85d2693bbf1" />

Running the server with debug level 2, it looks like when using the dialbox, the active channel is not being set.

When the transfer button is clicked:

127.0.0.1 -> Action: Atxfer
127.0.0.1 -> Channel: SIP/4605-00000138
127.0.0.1 -> Exten: 4008
127.0.0.1 -> Context: from-internal
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True

When using the dialbox:

127.0.0.1 -> Action: Atxfer
127.0.0.1 -> Channel:
127.0.0.1 -> Exten: 4008
127.0.0.1 -> Context: from-internal
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True

I'm not sure if this makes any difference, but we are running asterisk 1.6.2 and freepbx 2.8.0.2 in user/device mode.

Thanks for your help.

We are planning to use FOP2 for our reception staff. Their main role, as with many receptionists, is to forward calls to internal user extensions and queues. Used in conjunction with the presence information provided by FOP2 (which our existing solution does not have), this works well.

But, I cannot work out how to abort an attended transfer. This is required when somebody else picks up the target extension (physically or through a pickup group), or when the target chooses not to speak to the original caller and would prefer that reception put them straight through to voicemail.

The only way I can see to achieve this is to use the disconnect feature sequence (*0 or **), but that means switching back and forth between FOP2 and the phone. Is there a better workaround?