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

[Daniel Neubert]

The call fails because the desired gateway is down.

Logs are not available at the moment and issue cannot be reproduced on 
demand. I'll take logs as soon as this occurs again.

Best regards / Mit freundlichen Grüßen,
Daniel

On 28.07.2010 10:08, Steven Ayre wrote:
> Where & why does the call fail?
>
> Do you have any log file output?
>
> -Steve
>
>
>
>
> On 28 July 2010 08:25, Daniel Neubert <daniel.neubert at solomo.de 
> <mailto:daniel.neubert at solomo.de>> wrote:
>
>     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
>
>
>
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