In-Call codes

I am researching, why I cannot do any In-Call transfers. I have goggled around a lot.

I have VoIP phones, softphones, a mediatrix etc.

“In-Call Asterisk Attended Transfer” is set to *2 and enabled.
“In-Call Asterisk Blind Transfer” is set to ## and enabled.

Nothing happens; the “sound” is just transferred through to the caller. So you can hear it on the other phone.

I have also the “In-Call Asterisk Disconnect Code” and the “In-Call Asterisk Toggle Call Recording” set, they also do not work.

So, my installation does not react to In Call codes.

Any ideas on this?

So, for all of you call transfer just works fine ?

Am I the only one with trouble ?

In call transfer works fine.

You did not tell us what version of FreePBX, how it was installed (distro or manual), if manual what OS, version of Asterisk, what type of phones and anything else that would help us troubleshoot.

Thanks.

We run FreePBX 2.9.0.12 and Asterisk 1.8.7.1 on CentOS 5.5, installed as a distro.

Most phones are analog phones on a Mediatrix 4108, also Siemens Gigaset VoIP Phone and Acrobits softphone.

Which distro?

You mean what distro number?

I have no idea other than the version numbers above. How would I find out?

remember where you downloaded the distro from?

I say that because it smells of elastix, which cannot be reasobly supported here.

I thought it was a Freepbx distro. I inhearited th system, I was not the one who set it up.
How do I tell the difference?

When connect to the box with a browser to go from there to the admin interface, it says in the source code (and on the screen)

FreePBX-Distro

Are dtmfs working fine with those phones? Have you tried using a softphone/other phone?

Stupid q: Is *2 even the code or has the past sys guy changed it?

What does the full log say when you hit *2? Is *2

MikeK,

I am not sure if your still having issues with this, but I was searching on how to initiate call recording from end-user and just resolved the same issue you were having. Under the Setup Tab - General Settings - Dialing Options: Add T to the Asterisk Dial command options and Asterisk Outbound Dial command options.