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Not forwarding to cell phone anymore, used to work.

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saratis

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Previously I had this setup working just fine:

Incoming call->Digital receptionist->make choice->transfer call to ext 11->if unreg->transfer to 12->if unreg->transfer to 13->if unreg->transfer to 10 (call mobile)->no anwer->goto ext 82 (no one available)

11: 3CX phone on PC 1
12: 3CX phone on PC 2
13: 3CX phone on Android
10: My mobile phone (or thats what it's supposed to be)

Ext 10 was set up like this:
General->Mobile number: [my cell phone]
Forwarding->Available->No answer: goto ext 82
Forwarding->Available->Unregistered: goto ext 82

What used to happen was, it would try every extension and if none were available, it would just ring my cellphone. If I did not pick that up, it would goto menu 82.

But recently, probably after an update, it stopped working. The system refused to call my cell phone when ext. 10 was tried. What it does now, is go to menu 82 directly. I can get it to work partially if I set Forwarding->Available->Unregistered: Call mobile number. But now, when I don't pick up the mobile phone, it just disconnects. Not what I want.

Can you advise me on how to restore this previous behavior?
 
Log of a failed sequence:

18:15:45.607 [CM503005]: Call(6): Forwarding: Ext:Ext.82@[Dev:sip:[email protected]:40600;rinstance=c8aa0f701c735e8e]
18:15:45.606 [CM503016]: Call(6): Attempt to reach <sip:[email protected]:5060> failed. Reason: Not Registered
18:15:45.606 [CM503017]: Call(6): Target is not registered: Ext.10
18:15:45.605 [CM503010]: Making route(s) to <sip:[email protected]:5060>
18:15:45.604 [CM505003]: Provider:[Voys] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [VGUA-2355996m] PBX contact: [sip:144690002@3xxxxxxxxxxx:5060]
18:15:38.708 Currently active calls - 1: [6]
18:15:33.797 [MS211000] C:6.1:3xxxxxxxxxxx is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
18:15:27.890 [CM503007]: Call(6): Device joined: sip:[email protected]:40600;rinstance=7087e85768232f86
18:15:27.888 [CM503007]: Call(6): Device joined: sip:[email protected]
18:15:27.880 [CM505001]: Ext.80: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:[email protected]:5060]
18:15:27.880 [CM503002]: Call(6): Alerting sip:[email protected]:40600;rinstance=7087e85768232f86
18:15:27.686 [CM503025]: Call(6): Calling Ext:Ext.80@[Dev:sip:[email protected]:40600;rinstance=7087e85768232f86]
18:15:27.640 [CM503004]: Call(6): Route 1: Ext:Ext.80@[Dev:sip:[email protected]:40600;rinstance=7087e85768232f86]
18:15:27.639 [CM503010]: Making route(s) to <sip:80@3xxxxxxxxxxx:5060>
18:15:27.637 [CM505003]: Provider:[Voys] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [VGUA-2355996m] PBX contact: [sip:3xxxxxxxxxxx@3xxxxxxxxxxx:5060]
18:15:27.635 [CM503001]: Call(6): Incoming call from +3xxxxxxxxxxx@(Ln.10000@3xxxxxxxxxxx) to <sip:80@3xxxxxxxxxxx>
 
Well, the log indicates that the call came in, went to the IVR and was then routed to ext 10, and then to ext 82 (IVR). I will assume that the other extensions were unregistered. My guess is that you need to take a look at a) the IVR number associated to this routing to verify that it goes where you want it to first, then b) verify ext 10 and validate each status and the forwarding associated to it to include both internal and external call handling.

In many cases, when a call is forwarded to an external number and depending upon how that is accomplished, there is no way for 3CX to handle the call further. How is the call getting forwarded, through SIP or PSTN gateway? If a gateway, then it is highly unlikely that you will be able to use the desired routing as 3CX will consider the forwarding completed once the gateway has taken control. 3CX will consider the connection as being answered. If SIP, then possibly depending upon how the cell carrier and the SIP carrier negotiate and handle the call. A change by either could affect how this type of call is handled.

A wireshark capture might shed some light on what is being presented to the 3CX system so that you can see how the call is being
manipulated from the 3CX end.
 
Thanks.

That all sounds very reasonable, but this used to work! Was this a bug then? Or undocumented feature?

I believe the problem is the fact that extension 10 is unregistered, but that is always the case, as it does not link to a 3CX extension, but to my cell phone. What I think used to happen was that somehow ext 10 was registered, and it just started ringen my cell phone.
And then when I did not pick up, I went to ext 82.

Is there a way that I can get this to work again? E.g. how do I get ext10 to show as registered?
 
How was the call getting to ext 10, Gateway (GSM, PSTN, other?), Voip provider, etc.? What version of 3CX are we dealing with here? Were you using a softphone on your mobile?

The issue appears to be that as long as extension 10 is not registered, then there is no way 3CX will forward a call to same. I notice that you have 3CX for Android on ext 13. Are you certain that this was not also the case for your phone? I know of no way that a cell phone can be registered to a SIP server without some type of middleware or interface to make the two dissimilar networks compatible with one another.

Your partial success is exactly how I would expect it to work under the assumption that a PSTN gateway was employed to forward the call. The system handed the call to the PSTN network which then took control of the call. Cell phones are typically configured to take and answer a call either by someone physically answering or by voice mail box. Taking 3CX out of the equation, for the moment, it is likely that the same behavior would occur if you simply dialed your cell from a landline or other cell. It would ring until such time as you got bored and hung-up or the network would likely timeout and handup for you rathet than letting you tie up bandwidth. You also don;t indicate what type of behavior is expected if your cell is busy.

If you have a GSM gateway that supports call progress and is SIP compliant, then perhaps you might get the desired behavior.
 
The call was forwarded to Ext 10 by other extensions (unregistered softphones). Ext 10 had nothing registered to it, no softphone or whatever.

The call to my cell phone was made via a VOIP provider, it uses sip.voipgrid.nl. I have no PSTN devices.

I don't know what version of 3CX I was using before the update, but now I'm using 10.0.24018.2322 SP.6.

When the call to my cellphone was rejected (busy or not picking up), 3CX would pick it back up. I understand why this hard to believe, but it really is how it worked, as I tested it over 15 times when setting up the system.

I'm certain 3CX for android was off, as this did not work as expected (call quality on shaky wifi).

Do you see any way to restore this behaviour? Maybe use ring groups? I was so nice to be able to ignore the call on my cell and just have 3CX handle the call further...
 
The only way I can see is to have the call forwarded directly to the cell phone number of interest. This would be accomplished in the forwarding rules for ext 10. The functionality about taking back control in the event of a no answer or busy is most likely in the hands of your VoIP provider as you will note that 3cx does not have a timeout setting that is associated to a call that is forwarded to an external number.

I am not aware of any changes made to 3cx relative to call handing as it relates to the scenario you describe. You had indicated that perhaps ext 10 was registered somehow and the only way I can see this happening is with a soft phone somewhere in your network and then maybe the box for ring my cell and phone simultaneously was checked.
 
The call cell simultaniously is grayed out, could be that I had it checked before upgrading. I would consider buying a license if this would fix the issue.

I read on the forum that it could be possible to know for 3CX that the call to my cell was answered or not when a SIP trunk is used. Which is what I'm using.

So my guess is now: it used to work because a) The call mobile simultaniously option was checked and b) 3CX knew what was happening with the call because of the SIP trunk functionality.

Does this sound plausible?
 
Possibly. You could always download the demo version which provides full functionality for up to 2 calls , try it and then buy.
 
Possibly. You could always download the demo version which provides full functionality for up to 2 calls , try it and then buy.
 
Possibly. You could always download the demo version which provides full functionality for up to 2 calls , try it and then buy.
 
Possibly. You could always download the demo version which provides full functionality for up to 2 calls , try it and then buy.
 
Possibly. You could always download the demo version which provides full functionality for up to 2 calls , try it and then buy.
 
Was the version before the upgrade a full demo version? If so, the FREE version is missing some features.
Also, what about the Rebound option? Were you perhaps using that?
 
No I was just using the free version. Will think about trying the trial.

Thanks so far.
 
I have had problems sometimes, when setting diverts to outside mobile or fixed line numbers. Since these cases are usually after hours emergency numbers (e.g. callout to landlord due to a plumbing problem), they have to be 100% reliable ...
For this reason, I've dropped using the unreliable external number divert, and use an extension based divert instead ...

Set up a new extension. Configure it so that it has the external number you want to divert to entered as the user's mobile number.
On the Forwarding rules page - I set 'Send call to my mobile number' for both 'No Answer' and for 'Phone is busy'. Then of course, tick the box for 'Ring my extension and my mobile at the same time. I also tick the box to allow accepting more than 1 call at a time ... probably overkill in there - but it works!!

On the Extension's 'Other' page - select 'PBX Delivers Audio'. You can also put custom outbound caller ID on the extension number too - which is useful if you want to uniquely identify the call source on a [diverted to] mobile.

You'll see that this extension registers with 3CX - even though there is no phone associated with it. It's also easy to find (depending what you called it) on the Extensions Management page.

Then - just use that extension (rather than an external number), as your diversion target, and it works every time!

Graham Hill
Foxhall Solutions
 
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