Calls left on hold after a transfer

SinfoOne

Premier Customer
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Aug 8, 2022
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Hi everyone,

I have an on‑prem 3CX PBX, updated to version 20.0 Update 7 (Build 1080 Release).

Suddenly I’ve started experiencing this issue:

  • A call comes into our reception
  • The call is put on hold
  • The receptionist notifies the colleague who will receive the transferred call
  • The receptionist transfers the call to the colleague
  • The colleague receives the transferred call
  • The original caller, the one who called reception, remains on hold, as if the transfer never happened
Nothing should have changed in the configuration, and I’m unable to find the cause of the problem.
Can you help me?

Thanks,
Luciano
 
Hi Luciano, is this just with the external calls or also with the internal calls?

if it is only with the external calls, what voip provider are you using?

what endpoints are they using, if they are deskphones, what provisioning method, firmware version and template are they using?

one last thing is this a blind or attended transfer?
 
Hi Alejandro,

thank you for your kind reply.

I’ve run several tests and found that:
  • from an internal call, transferring to an internal number = works
  • from an internal call, transferring to an external number = works
  • from an external call, transferring to an internal number = does not work
  • from an external call, transferring to an external number = does not work
My provider is BTEnia (now Retelit).
The endpoints performing the call transfers are Fanvil X7 devices running firmware 2.4.19; provisioning is done from the admin portal by selecting the user, “Phone Provisioning,” MAC address, etc.

We are using the Attended Transfer method.

Thanks,
Luciano
 
So, you should start looking at the trunk side, as first step I do not set that provider as supported, so you should be using a generic sip trunk template, if you download your current sip trunk do you see in the .xml file that support reinvite and replaces are enabled? those fields should not be used in the trunk side.

Please give it a try using a supported sip trunk and let us know if the issue persists.
 
Hello Luciano,

I will send you a p.m. shortly.
 
Hi Alejandro,

I checked, and in the XML file I see this:

IsSupportReinvite 1
IsSupportReplaces 0

Is one of these two values causing the problem?

Thanks,
Luciano
 
Hi Alejandro,

I checked, and in the XML file I see this:

IsSupportReinvite 1
IsSupportReplaces 0

Is one of these two values causing the problem?

Thanks,
Luciano
Yes, both should be set as 0

Edit the file using notepadd++, then from the advanced menu ---- templates ----- voip provider template upload the file.

But using the real generic sip trunk those fields are set as 0, it means this trunks was edited in some way or used a different template...
 

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