Hello, thanks for reading this post! Ok, so I’m pretty convinced this question has been answered a few times but I can’t find it so here goes…
I’ve got a medium sized system with roughly 150 phones and a call center that takes calls from a couple different queues. The problem that I’m having is when a call gets transferred to a queue from an internal user, two calls end up in the inbound calls list in iSymphony: the call that was supposed to end up there as well as a call from the transferring user. The actual call will eventually get answered and terminate just fine however the call from the person who transferred it sits in the queue until iSymphony is restarted. I would think this is a problem with iSymphony and I think I’ll open a case with them too however when transferring using ## the condition does not occur.
I suspect this has something to do with the behavior of SIP when transferring from Polycom phones. Maybe it does a reinvite instead of a refer or something like that but I’m really at a loss. I haven’t dove into packet captures yet because even if I uncovered this was the case I wouldn’t know what to do with that information.
Is there a way to edit the basefile such that the transfer button issues a ##? Or something like that?
Strangely this behavior was not present before a recent software upgrade. I don’t even know what to say about that because it was messy, I’m not even sure I can tell you what version I was on prior to the upgrade. I inherited this system and the previous administrator had attempted a 12-13 upgrade which had failed. Half the system was updated and thought fwconsole was the man, but amportal was still in use and fwconsole didn’t exist; at least not in the right location. I ran the manual upgrade script (among a lot of other stuff I had to do to nudge it along) and I believe the system to be completely up to date and healthy now.
Currently the system reports as running:
FreePBX 13.0.190.11
Current Asterisk Version: 13.13.1
I’m using Chan_SIP everywhere but I’ve enabled the PJSIP driver and wonder if that could be the fix?? I’ve not been able to get a PJSIP extension to register but I’m still RTFMing that.
Any direction you can provide would be appreciated!!