Hey guys - first of all, thanks for all the great help on here so far. This forum has been immensely helpful in my learning and our setup process.
I just setup a VPN for remote users to be able to use Zoiper softphone from their mobile devices to call out from our PBX. It works great!
The problem is, once the user disconnects, their respective deskphone is not registered anymore (although the Yealink web UI says it is). I have to go into the web UI and re-register it for the deskphone to accept calls. Otherwise, a caller is just dumped to that user’s voicemail.
Is there a better way to implement this? Giving each user 2 extensions doesn’t seem like the most efficient way, but it would work. Thoughts?
PJ-SIP allows for multiple instruments per line. That handles the “getting the call to the destination”. I don’t know about the desk phone registration thing, but one would assume that PJ-SIP knows how to handle that.
If you aren’t using PJ-SIP (I don’t need a reason, just if), you can use Find Me/Follow ME and point the desk phone to the cell extension. The tricky part with this is you have to make sure the cellphone’s timeout for Voicemail is longer than yours, otherwise your business voicemails end up in the employee’s personal voicemail box. This method relies on the desk phone having one extension and the cell phone having another. The advantage with this approach is that the cell phone rings only when a call is being sent to it. Judicious use of extension numbers and Caller ID can give this the appearance of a pair of phones working together.
The only disadvantage here is that the customer may have to sit on ‘ring’ while the call is placed. I usually use an announcement (“Please wait while I connect your call.”, IIRC) to let people know that the call is still being handled.
This second method is a “tried and true” architecture that has served many of my customers very well over the years. The first is new, and should be useful in a few cases.
Thanks, really appreciate the ideas. I decided to give PJSIP a shot first, but when I dial out of a PJSIP test extension to either an internal or external number, the PJSIP phone can’t hear the other device. The other device (SIP or external phone) can hear me just fine. This also applies to voicemail boxes and our conference rooms - if I call them from the PJSIP phone, I hear nothing. Everyone else hears me.
IIRC, the audio part of a SIP call (regardless of channel driver) uses port 10000-20000. These need to be open and redirected to your PBX for the calls from outside the network to connect. Interior calls should work fine, since internal calls all know how to get to the server.
That’s the weird thing, the interior calls are having the same issue. I tried these test calls:
PJSIP extension -> SIP extension - SIP could hear me, I can’t hear SIP
PJSIP -> conference room - I can’t hear anything, but I was joined to the room
I tried this with
firewalld disabled on the FreePBX server too, same result.
I believe you’d solve your problems (and gain great flexibility) by switching to Device & User Mode (and sticking with CHAN-SIP). Register all hard- and softphones as Fixed devices and assign the appropriate User.
Alternatively, look into DISA (in the Applications menu) - I’ve never played with it myself but it looks relevant.
In device and user mode, are all devices still assigned new extensions?
Ahh, nevermind, this should help me out: http://wiki.freepbx.org/pages/viewpage.action?pageId=5242941