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?