<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
The call fails because the desired gateway is down. <br>
<br>
Logs are not available at the moment and issue cannot be reproduced on
demand. I'll take logs as soon as this occurs again.<br>
<pre class="moz-signature" cols="72">Best regards / Mit freundlichen Grüßen,
Daniel
</pre>
On 28.07.2010 10:08, Steven Ayre wrote:
<blockquote
cite="mid:AANLkTinLBUair+T=HvhNn4fdvawnyeRL7t+BtiQ6qwHO@mail.gmail.com"
type="cite">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 moz-do-not-send="true"
href="mailto:daniel.neubert@solomo.de">daniel.neubert@solomo.de</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; 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>
_______________________________________________<br>
FreeSWITCH-users mailing list<br>
<a moz-do-not-send="true"
href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a moz-do-not-send="true"
href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users"
target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a moz-do-not-send="true"
href="http://lists.freeswitch.org/mailman/options/freeswitch-users"
target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a moz-do-not-send="true" href="http://www.freeswitch.org"
target="_blank">http://www.freeswitch.org</a><br>
</blockquote>
</div>
<br>
</blockquote>
</body>
</html>