Hi, we've set up a SIP trunk between Asterisk (used as MediaGateway to SS7-Network for PSTN access) and Freeswitch. Everything works fine except one "little" issue: If there have been no calls using the SIP trunk it becomes unuseable from Freeswitch side. PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -> VoIP Clients If a user tries to originate the call that is routed to one of our MediaGateways while SIP trunk is "stale", the call will fail. The trunk can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk This is our gateway configuration (tried using low values for expire-seconds, ping and retry-seconds to keep the connection up: <gateway name="voip-int-test"> <param name="username" value="voip-ext-test"/> <param name="password" value="freeswitch"/> <param name="proxy" value="172.31.45.43"/> <param name="register" value="false"/> <param name="expire-seconds" value="15"/> <param name="ping" value="5"/> <param name="retry-seconds" value="5"/> <param name="context" value="default"/> <param name="apply-inbound-acl" value="voip-int-test"/> <param name="caller-id-in-from" value="true"/> </gateway> -- Best regards / Mit freundlichen Grüßen, Daniel