Voicemail dropping calls

asterisk engine, running one sip trunk to the pstn
I have a problem with voicemail dropping/disconnecting incoming calls on the sip trunk. I have incoming calls going to extension 23. If I dial ext 23 from another extension the voicemail works fine, but if a call comes in from outside (on the sip trunk) there is instant disconnection of the call when the voicemail kicks in. I have voicemail switched off at my sip provider so that asterisk catches the call. I tried changing the incoming call to another extension but it still drops just the same. I’ve watched the debug in a console during an incoming call but all it says is ‘Request Cancelled’ and doesn’t give any reason. What might cause this disconnection?

“Request Cancelled” on an incoming SIP INVITE means the caller abandoned the call before it was answered. The cancel is initiated by the side that sent the INVITE.

thanks for your response David, that helped and based on what you said I tried a temporary switch to a different sip provider and whaddyerknow… incoming calls work fine with no more line drop. So it looks like the sip provider’s machinery is at fault and I’ve written to them and hopefully they will correct any problem at their end.