If running your asterisk server behind a firewall you may experience these symptoms:
internal users can call mobile extensions configured with follow me and the call goes through and can be heard with no problem.
If you have a rule configured so that external callers after a timeout calling internal extensions are then routed to call an external mobile, the call rings the mobile cell phone and can be answered but no sound can be heard either way.
First make sure you have the required RTP and SIP ports forwarded to the internal VoIP server.
For asterisk this typically means having port 10000-20000 and 5060 forwarded.
Edit /etc/asterisk/sip_general_custom.conf and add the following entries where EXTERNALIPADDRESS is the Public IP of your internet.
Go to http://whatismyipaddress.com to check
Locanet is the local LAN subnet the VOIP server is on
If using elastix you can edit the sip_general_custom.conf file under the PBX menu -> Tools -> Aserisk File editor.
Filter for sip_general_custom
Open the file, copy and paste the above and change the EXTERNALIPADDRESS.
Save the file
Test the redirection to mobile