Flowroute as SIP provider.
My client is having issues with leaving voicemail on external phone numbers getting clipped around the 10s point. I was curious if there is something I am missing to remedy this.
In Asterisk SIP Settings, try setting RTP Timeout to 180 (3 minutes). If no luck, paste a log of a failing call, including SIP trace, at pastbin.freepbx.org and post the link here.
Check and see if AMD, Answering Machine Detection is enabled. If so, try disabling it. It is a module. My nephew had a similar problem when leaving messages on his FPBX and this fixed his issue.