Transferring calls internally

hi Guys,

any help would be appreciated. I am very new to asterisk and trixbox so i am not very knowledgeable.

We are currently having an issue where once we attempt a transfer the call starts to beep and then disconnects.

I am not entirely sure where to start with the diagnostics of this so any help would be greatly appreciated. If you require any more details please let me know.

Trixbox has been a dead product for some time now, so it should not be possible to be new to it.

Enable full logging in logger.conf.

Do core set verbose 5 and core set debug 5 at the CLI prompt.

Enable the channel driver debugging for the channel with the problem (you didn’t say which). Look through the output for obvious errors.

You will also need to tell us how you are attempting the transfer, as both SIP and dahdi allow two basic classes as well as allowing attended and unattended ones. At least some SIP phones do everything as attended.