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@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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