[Freeswitch-users] Keep alive for SIP trunk between Asterisk and Freeswitch

[Daniel Neubert]

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