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

[Steven Ayre]

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> 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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