[solved] Transfer and moh not working, please help

[MrWeb]

Hello @ all.

i have a crazy problem in my FusionPBX and i hope, someone can help
me to solve this.

At the first, a short description.
I setup a new Debian 10.11 with FusionPBX as a test maschine for configuration.
Here i setup all my extensions, routes and so on. I tested the
provision of the phones and all the functions.
For testing, i use a second fusionpbx server as gateway
(with the magic line from DigitalDaz) so, the phone on the gateway (second fusionpbx)
act as the line side. That works perfect. So i can call in, like from the provider.
At this point, all is ok.

Now, i transferd all to my production machine.
All the same, complete same setup, ip adress and so on.

The only differences are:
- i set the externel IP Adresses for
external_rtp_ip and external_sip_ip

- i changed the config of the gateway to the real provider
(german telekom sip-trunk)


I can call in, call out. All work like configured.
BUT... and now my problem(s):

(I use only yealink phones T23G / T46S/ T48S)

If i press the button 'Hold', the call is hold, but the caller
hears no music on hold. I can get back the call, if i press
the button again. So, the button is working, but no moh.

And the second, bigger problem. If i press the button for 'Transfer'
the call is hold, and the phone starts a new call to the destination.
No transfer/bridge is possible. Also, the caller hears only silence,
no moh. i can get back the call, with the hold button.
No transfer is possible :-(

On the test maschine where i configured all, both is working there.
Moh and the transfer (with moh)

I do not understand, where the problem is here.
I changed only the gateway.
The logs show no errors. If i compare the logs from
testmashine to production, there is not really a differenece.

The transfer did not work for internal call and for external also.

I hope someone of you can give me some information to fix that.

If you need more informations, let me know.

Thanks to all for reading this.

Best regards,
MrWeb

EDIT:
I atttached two log files, one from production system, the other from the testsystem.

testsystem is working
production system not

For both log files:
I call from extension 600 to extension 602 (both internal)
After 5 seconds, i press the hold button on 602.
production: -> silence on 600
testsystem: -> moh on 600

After 10 seconds, i press the hold button again.
I get the call on both systems back.
But if i hangup on 602
productionsystem: -> call open, i need to hangup myself the call
testsystem: -> call clean closed

So, i hope someone can see somthing, what is going wrong on production system.
domain is: pbx1.mydomain.de on both
siptrunk base is: +49|0 1234567 on both

Best regards
MrWeb

 

[flagman]

Hello,
Can you please tell me the yealink Firmware version? I had the same problem with one version of the firmware. it was solved after an update.
Also please check the transfer settings in Yealink GUI, (Features>Transfer)

 

[MrWeb]

Hi and thanks for your reply.
Yealink T23G FW 84.0.140 as example.
All Phones uses the same, newest firmware.

In my case, the firmware could not be the problem.
I use the same phones for my internal testsystem, here all is working
and in the productive system, it is not working.
Feature->Transfer is also the same config on both.

Any other ideas ?

 

[bcmike]

Hi and thanks for your reply.
Yealink T23G FW 84.0.140 as example.
All Phones uses the same, newest firmware.

In my case, the firmware could not be the problem.
I use the same phones for my internal testsystem, here all is working
and in the productive system, it is not working.
Feature->Transfer is also the same config on both.

Any other ideas ?

I know you say its not your firmware but try rolling it back anyway. Same happened to me.

 

[MrWeb]

Ok, i solved the problem.
The firmware was not the problem.

NAT was the problem. I changed the remote_sip_ip and remote_rtp_ip to the external.
But in this case, forwards and so on, did not work anymore.
So i changed both back to the internal ip.
As next, i changed in the external sip profile the value
ext-rtp-ip
and
ext-sip-ip

to autonat:EXTERNALIP

and with this changes, everything is no working

Thanks