Where & why does the call fail?<br><br>Do you have any log file output?<br><br>-Steve<br><br><br><br><br><div class="gmail_quote">On 28 July 2010 08:25, Daniel Neubert <span dir="ltr"><<a href="mailto:daniel.neubert@solomo.de">daniel.neubert@solomo.de</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Hi,<br>
<br>
we've set up a SIP trunk between Asterisk (used as MediaGateway to<br>
SS7-Network for PSTN access) and Freeswitch.<br>
<br>
Everything works fine except one "little" issue: If there have been no<br>
calls using the SIP trunk it becomes unuseable from Freeswitch side.<br>
<br>
PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -><br>
VoIP Clients<br>
<br>
If a user tries to originate the call that is routed to one of our<br>
MediaGateways while SIP trunk is "stale", the call will fail. The trunk<br>
can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk<br>
<br>
This is our gateway configuration (tried using low values for<br>
expire-seconds, ping and retry-seconds to keep the connection up:<br>
<br>
<gateway name="voip-int-test"><br>
<param name="username" value="voip-ext-test"/><br>
<param name="password" value="freeswitch"/><br>
<param name="proxy" value="172.31.45.43"/><br>
<param name="register" value="false"/><br>
<param name="expire-seconds" value="15"/><br>
<param name="ping" value="5"/><br>
<param name="retry-seconds" value="5"/><br>
<param name="context" value="default"/><br>
<param name="apply-inbound-acl" value="voip-int-test"/><br>
<param name="caller-id-in-from" value="true"/><br>
</gateway><br>
<br>
<br>
<br>
--<br>
<br>
Best regards / Mit freundlichen Grüßen,<br>
Daniel<br>
<br>
<br>
<br>
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</blockquote></div><br>