RTP Keepalive not written to 'pjsip.endpoint.conf' file (pjsip only)

When an incoming external call is forwarded (or FMFM:ed) on Freepbx to an external party and the provider is expecting to receive RTP, before sending any RTP, there is no voice connection (caller and callee cannot hear each other).

A possibility to address this problem is to configure 'rtp_keepalive=X' (X>0) in 'Asterisk SIP Settings'. However Freepbx does not appear to write that value to any pjsip configuration file, in particular not to 'pjsip.endpoint.conf'.

The attached PHP code patch should resolve this issue.